| 00:03.06 | p3nguin | Sounds messed up to me. |
| 00:03.53 | p3nguin | I'm replacing your scenario with an SMTP/email situation... I'm trying to imagine if the smtpd sent email back to the source port instead of to the standard smtp port. |
| 00:08.07 | Tuju | http://forums.whirlpool.net.au/archive/1161875 there's another similar case, with no solution. |
| 00:09.44 | nny | doe originate wait for the channelt obe answered before running an app? |
| 00:10.48 | p3nguin | yes |
| 00:11.14 | p3nguin | The channel you're connecting "from" has to go Up before it will run an app or extension. |
| 00:11.15 | nny | k, i think my issue is when I am calling originate it's not passing channel variables, which make sense since it's a new channel |
| 00:11.56 | p3nguin | Create a dial plan that sets your stuff, then call a Local channel against that dedicated piece of dial plan. |
| 00:12.27 | nny | p3nguin: this variable is passed from the original channel, in this case the variable is a meetme room. I think I can reverse the process though and make my local channelt he noise I want to play |
| 00:12.42 | nny | it doesn't exist outside the channel |
| 00:14.16 | nny | could pass it to a unique global if I needed to... testing this first |
| 00:24.06 | nny | Arrrrggh! |
| 00:24.07 | nny | lol |
| 00:24.16 | nny | so I am trying to connect two users in a meetme |
| 00:24.34 | nny | one connects with exten => s,n,Meetme(1234,cdaA) |
| 00:24.56 | nny | the other, Meetme(1234,q) |
| 00:25.13 | nny | i want the first one to hear the announce, leave, but not the second... |
| 00:25.16 | nny | it doesn't work |
| 00:25.40 | nny | however if I use a local channel to connect to the meetme room to play a sound, and use the same option as user 1, i hear the enter leave |
| 00:25.56 | nny | so in the process of playing the sound, it plays the "enter, leave" noiose |
| 00:25.58 | nny | noise* |
| 00:26.01 | nny | which confuses me |
| 00:26.23 | nny | since the local channel uses the same exact connection string as the.. oh.. hmm |
| 00:26.37 | nny | could this be how the original channel was dialed (call file?) vs the local channel? |
| 00:27.16 | p3nguin | Local channels are intended to be optimized out of the call. If you want to keep it in the call, I think you need to append /n to the end of the Local channel that you're calling. |
| 00:27.25 | nny | er |
| 00:27.31 | nny | p3nguin: so what should happen here? |
| 00:27.45 | nny | p3nguin: 1 user calls with cdaA, 2 with q to Meetme |
| 00:27.55 | nny | p3nguin: should user 1 hear user 2 join? |
| 00:28.00 | nny | (and leave) |
| 00:28.14 | p3nguin | I have no idea without looking at the options. |
| 00:28.29 | nny | c is create, d dynamic, Aa is admin modes |
| 00:28.32 | nny | and q is quiet |
| 00:28.38 | nny | punches a baby |
| 00:29.06 | nny | i have been trying to get a simple "don't alert user 2 in meetme" to work in various ways, and the damn thing is suppose to work by default, but is being killed by something else |
| 00:30.54 | nny | I am using a call file to generate the call, maybe that has something to do with it.. although w/e it is it's not inuitive. |
| 00:30.58 | p3nguin | Your idea is that if caller1 is in a meetme, regardless of the options used, if caller2 is put into a meetme with q option, caller1 won't hear the sounds? |
| 00:31.07 | nny | no, that's the problem |
| 00:31.12 | nny | my idea is it SHOULD work as intended |
| 00:31.40 | nny | i.e. caller 1 hears caller 2 announce, but caller 2 doesn't hear it.. by default when you join a conference you hear the ba-dee-beep noise |
| 00:31.54 | nny | but "q" should prevent that |
| 00:32.13 | nny | however it just kills the entry noise all together OR the noise isn't being played to the channel due to call file being used or something |
| 00:32.15 | p3nguin | If caller1 is in the meetme and you didn't use q, and caller2 enters meetme using q... |
| 00:32.20 | nny | yes |
| 00:32.23 | nny | it should work |
| 00:32.25 | p3nguin | If caller1 leaves, do you expect caller2 to hear the sound? |
| 00:32.28 | nny | yes |
| 00:32.37 | p3nguin | If caller2 leaves, do you expect caller1 to hear the sound? |
| 00:32.37 | nny | just want to prevent caller 2 from hearing either |
| 00:32.43 | nny | er sorry |
| 00:32.50 | nny | if caller1 leaves, caller2 should hear nothing |
| 00:32.56 | nny | if caller2 leaves, caller 1 should hear a sound |
| 00:32.57 | *** part/#asterisk high-rez (~gus@carrera.bourg.net) |
| 00:33.09 | p3nguin | I think you'd need caller1 to use the q option, then. |
| 00:33.38 | p3nguin | and caller2 should not use q. |
| 00:33.43 | nny | .... |
| 00:33.48 | nny | hahaha |
| 00:33.50 | nny | i'll try it |
| 00:35.48 | nny | p3nguin: well.. almosyt |
| 00:36.09 | nny | p3nguin: except caller 2 can hear his own entry (if only for a brief sec) when he joins |
| 00:36.28 | nny | guess my explanation sucks |
| 00:36.32 | nny | that was the original issue |
| 00:37.06 | p3nguin | If you are in a meetme, I should be able to enter using the q option and you won't hear me come in. |
| 00:37.15 | p3nguin | regardless of which options used when you went in. |
| 00:37.23 | p3nguin | At least that is my interpretation of the options. |
| 00:37.40 | nny | p3nguin: the issue is this: Caller 2 can hear himself join through the meetme application |
| 00:38.04 | p3nguin | If caller2 does not use q, I'd expect that. caller1 will also hear c2 join in. |
| 00:38.08 | nny | p3nguin: right |
| 00:38.19 | nny | p3nguin: was trying to work around it.. i have some ideas. |
| 00:38.35 | nny | p3nguin: like local channel connecting before and playing a sound, leaving, then caller 2 joins.. |
| 00:39.17 | p3nguin | Can the admin change the options of a caller during the conference? |
| 00:39.34 | nny | p3nguin: some what |
| 00:40.09 | WIMPy | Good morning |
| 00:40.50 | nny | wonder if I can change the noise the meetme app uses |
| 00:41.15 | nny | that would solve it. the ba-dee-dee noise is just too long, it hits the channel in the brief time before caller2 does |
| 00:42.37 | nny | can use marked mode I suppose and some moh too. OK enough of that, just have to work with what I have |
| 00:44.14 | nny | new question, the h extension runs until there's nothing more to do? |
| 00:45.04 | p3nguin | The h extension runs just like any other extension. It starts with priority 1 and executes each priority until there isn't another to execute. |
| 00:45.21 | nny | p3nguin: yeah, but it does so after the channel has been hungup |
| 00:45.35 | nny | p3nguin: however it seems to be dying earlier than I expect, maybe something to do with my Wait() command |
| 00:45.42 | p3nguin | That's what h means: hangup |
| 00:46.23 | WIMPy | Not all dialplan apps work in h. |
| 00:46.50 | nny | WIMPy: yeah so I've noticed, is there a list? |
| 00:47.09 | WIMPy | I don;t know of one. |
| 00:47.21 | nny | WIMPy: audio channel stuff for obvious reasons.. but wait? is it a timing issue? |
| 00:47.23 | p3nguin | Anything that uses a channel shouldn't work. |
| 00:47.30 | WIMPy | But I'm pretty sure, Wait would be on it. |
| 00:47.44 | nny | haha figures |
| 00:47.49 | nny | hack hack hack hack |
| 00:47.57 | nny | all i've done the last two days |
| 00:48.09 | WIMPy | Days? |
| 00:48.13 | nny | :P |
| 00:48.14 | WIMPy | Hah. |
| 00:48.17 | nny | well |
| 00:48.21 | nny | in recent history |
| 00:48.26 | nny | as in this week :\ |
| 00:49.45 | nny | WIMPy: lol wonder if I can use a local channel to call the wait |
| 00:49.49 | nny | :\ |
| 00:50.01 | WIMPy | Probably not. |
| 00:50.51 | Tuju | p3nguin: see http://www.voip-info.org/wiki/index.php?page=Standalone+Cisco+7941%2F7961+without+a+local+PBX at the end of the page |
| 00:51.09 | nny | WIMPy: like Originate(s/@wait-local/n,app,Wait,5) or evebn bridge it to another local channel? |
| 00:51.35 | WIMPy | Originate shouldn't wait. |
| 00:52.02 | WIMPy | But I haven't tried that. |
| 00:52.11 | Tuju | ouh crap, how those ciscos can be so difficult |
| 00:52.24 | Tuju | i need sleep |
| 00:52.26 | nny | er Local/s* |
| 00:52.29 | nny | trying it. wth |
| 00:52.34 | Tuju | gonna suspend myself into charger. |
| 00:52.43 | WIMPy | Tuju: No, you just use them in the wrong way. |
| 00:52.56 | Tuju | yeah, landfill, i know. |
| 00:53.08 | WIMPy | That's an option. |
| 00:53.18 | WIMPy | But I gues they would work with CCM as well. |
| 00:53.23 | p3nguin | I don't know what part you're wanting me to read. |
| 00:53.28 | Tuju | sounds like CCCP |
| 00:53.41 | Tuju | end of the page |
| 00:53.53 | p3nguin | Yeah, you mentioned that. |
| 00:54.01 | Tuju | Configure port mapping on gateway router |
| 00:54.07 | Tuju | that explains the problem |
| 00:54.45 | Tuju | i got it once working with commercial gw provider, which probably was caused by fact that they do that port remapping already |
| 00:54.54 | p3nguin | That's not at the end, that's in the middle. I don't know how you expected me to GUESS that's what you wanted me to read. |
| 00:55.20 | Tuju | i'm tired, going to sleep now. |
| 00:55.30 | p3nguin | I know the feeling. |
| 00:55.48 | nny | WIMPy: ding you win.. sigh lol |
| 00:55.59 | nny | WIMPy: yeah Originate just pops off, jumps to next part of dialplan |
| 00:56.02 | nny | hmm |
| 00:56.12 | nny | need 5 seconds.. may have to make a loop or some garbage |
| 00:56.15 | WIMPy | That's what I expected it to do. |
| 00:56.27 | nny | WIMPy: indeed, I just wanted to fail all by my lonesome |
| 00:57.36 | nny | oh voip-info,.. Wait could be modified to work on a dead channel in extension h, but as it stands right now the code assumes the channel is alive, and to ensure that things work properly, it has to read incoming media from the channel so it can be discarded... and if at any time waiting for or reading media from the channel fails, it exits, because there's no point in continuing to wait since the call is gone. |
| 00:57.36 | WIMPy | smells some desperation |
| 00:57.57 | nny | WIMPy: haha, well.. i'm 95% done, this is just some post call code... |
| 00:58.35 | nny | WIMPy: hahaha System should do it |
| 00:58.56 | WIMPy | Yes, That should work. |
| 00:59.01 | nny | WIMPy: System(sleep 5) |
| 00:59.02 | nny | ? |
| 00:59.04 | nny | hahaha |
| 00:59.06 | nny | sigh |
| 00:59.26 | p3nguin | Include the full path of sleep. |
| 00:59.45 | nny | p3nguin: ahh yes. Thanks |
| 00:59.48 | *** join/#asterisk gravin (~gravin@204.34.49.60.brf01-home.tm.net.my) |
| 01:02.41 | nny | perfect |
| 01:02.42 | nny | lol |
| 01:03.03 | nny | some day i'll have to share my dialer code, and you can all laugh at me for not using C or something |
| 01:03.21 | nny | although i was told by others scripting languages work just as well |
| 01:03.54 | nny | it's going well, almost fun at times |
| 01:03.56 | nny | almost |
| 01:06.44 | WIMPy | Hmm. AMI is cool. |
| 01:07.12 | WIMPy | And I certainly wouldn't say scripting works as well. Not if it gets komplex. |
| 01:07.59 | nny | WIMPy: yeah for the most part it's not. Just creating call files and connecting them to meetmes. I don't prefer the latter (queues or something) but it's coming along well |
| 01:08.12 | WIMPy | Well. It's not always cool, but it allows quite some control. |
| 01:09.03 | nny | WIMPy: yeah |
| 01:09.44 | nny | this is essentially a loop cycle, with sanity checks and breaks... + some database management |
| 01:10.12 | *** join/#asterisk di0_ (~di0_@bd218666.virtua.com.br) |
| 01:10.31 | nny | create meetme room -> create call file -> AMD -> create another -> HUMAN -> connect to meetme room -> (TALK) -> create call file etc |
| 01:10.40 | WIMPy | I used AMI DBGet the forst time about a fortnight ago and found tha a little over complicated. |
| 01:10.51 | *** part/#asterisk di0_ (~di0_@bd218666.virtua.com.br) |
| 01:11.18 | nny | WIMPy: this is either odbc stored procedures called from dialplan or bash script importing values from a local db (mysql) |
| 01:11.24 | WIMPy | But you use it for more that 2 channels? |
| 01:11.37 | nny | WIMPy: and really it's like Set(ODBC_SOMETHING=${STUFF}) |
| 01:12.02 | nny | WIMPy: yes each meetme room "created" (disguised as an agent login) creates the call files directly for that meetme |
| 01:12.18 | nny | WIMPy: and during it constantly checks the occupants of the room |
| 01:12.22 | nny | to avoid awkward situations :D |
| 01:12.41 | WIMPy | Sounds interesting :-) |
| 01:13.01 | nny | WIMPy: heh it started as "build me a bike" and ended up "build me a bike, track, tires and make them square" |
| 01:13.21 | WIMPy | Yes, as usual. |
| 01:13.25 | nny | :) |
| 01:14.10 | WIMPy | Althoug the usual is more like: I need a bike. Ok, here's your bike. And where is the road for that? |
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| 01:14.36 | nny | WIMPy: heh |
| 01:15.43 | *** join/#asterisk coppice (~chatzilla@116.92.39.71) |
| 01:26.10 | james_zhu | hi |
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| 02:21.55 | bougyman | vici users? I'm following the doc and it says zaptel-1.4.12 but that won't build against my kernel, even the -lts version (2.6.32 on arch linux) |
| 02:22.04 | bougyman | anyone know if it's ok to use dahdi instead? |
| 02:23.50 | WIMPy | dahdi is the new name of zaptel. Yes, you have to use dahdi with not too old versions of Linux and/or Asterisk. |
| 02:28.33 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 02:30.14 | jeffspeff | how can i get the extension of a user for use in a context? example--> exten=s,n,VoiceMailMain(${EXTEN}@star) <--- but that doesn't work because it returns ${EXTEN} as "s" |
| 02:31.01 | WIMPy | Callerid? If it's the same. |
| 02:31.35 | p3nguin | How does the call get to that context/extension in the first place? |
| 02:31.49 | p3nguin | I use a preset variable from earlier in the dial plan. |
| 02:31.53 | jeffspeff | exten=*25,1,Goto(star-own-vmail,s,1) |
| 02:32.52 | jeffspeff | i guess i could set callerid(num) for each user, and then call that in voicemailmain() |
| 02:33.30 | p3nguin | VoiceMailMain(${CALLERID(num)}@star) |
| 02:33.58 | jeffspeff | yeah, now just to go back and set the cid for 40 people... yay! |
| 02:34.04 | jeffspeff | thanks everybody |
| 02:34.08 | p3nguin | or |
| 02:34.09 | p3nguin | exten => *25,1,Set(__TargetNum=${CALLERID(num)}) |
| 02:34.11 | p3nguin | exten => *25,n,Goto(star-own-vmail,s,1) |
| 02:34.33 | p3nguin | And then exten => s,1,VoiceMailMain(${TargetNum}@star) |
| 02:34.49 | p3nguin | either way would be pretty much the same end result. |
| 02:34.54 | jeffspeff | and __TargetNum fetches the callerid? |
| 02:35.05 | WIMPy | They don't have callerid? |
| 02:35.20 | p3nguin | That first line sets the var TargetNum to the caller id number. |
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| 02:35.51 | jeffspeff | their callerid is their name... |
| 02:36.08 | p3nguin | That's CALLERID(name) |
| 02:36.09 | jeffspeff | i haven't toyed with setting both yet... i'm using realtime ldap |
| 02:36.10 | p3nguin | We're talking about CALLERID(num) |
| 02:36.14 | jeffspeff | right |
| 02:36.22 | jeffspeff | i only have name set so far |
| 02:36.42 | p3nguin | You're trying to say that the phones don't have CALLERID(num)s? |
| 02:36.55 | p3nguin | I'd be surprised if that's right. |
| 02:37.16 | jeffspeff | not internally within the office... i set the callerid num on outbound calls |
| 02:37.47 | p3nguin | I'll be really surprised if phones don't send something with every single call, no matter what. |
| 02:39.06 | jeffspeff | the sip users are all user names... the call comes in as being from that username... when you try and return a missed call or something that was stored on the phone from a previously received call, the phone dials the sip user name, not the extension |
| 02:39.42 | jeffspeff | i had to add a "exten=kspiess,1,Goto(602,1)" for each user so they could call each other back |
| 02:39.58 | jeffspeff | <PROTECTED> |
| 02:39.58 | jeffspeff | <PROTECTED> |
| 02:39.58 | jeffspeff | <PROTECTED> |
| 02:39.58 | jeffspeff | <PROTECTED> |
| 02:40.07 | jeffspeff | my bad on, the multi... |
| 02:40.11 | p3nguin | You know that your syntax is broken, right? |
| 02:40.14 | jeffspeff | thought it was all going to be one line |
| 02:40.22 | jeffspeff | p3nguin, what part? |
| 02:40.32 | p3nguin | exten => 602,1,stuff() |
| 02:41.09 | p3nguin | Note the spaces and the angled bracket. |
| 02:41.17 | jeffspeff | that's an old copy paste from a dialplan i did a long time ago... it doesn't error out, so i've never bothered to mess with it |
| 02:41.45 | p3nguin | I see |
| 02:42.11 | jeffspeff | there's not an angled bracket |
| 02:42.37 | p3nguin | exten => 602,1,stuff() <--- there should be |
| 02:52.25 | ChannelZ | parenthesis |
| 02:52.43 | p3nguin | hmm? |
| 02:53.35 | ChannelZ | () |
| 02:53.41 | p3nguin | Yeah? What about them? |
| 02:54.03 | ChannelZ | ...nevermind |
| 03:01.29 | *** join/#asterisk gravin (~gravin@175.136.216.20) |
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| 03:24.32 | *** join/#asterisk nobodyshome (~nobodysho@bas10-kitchener06-1279411550.dsl.bell.ca) |
| 03:24.59 | nobodyshome | p3nguin, ya there |
| 03:25.31 | nobodyshome | or anyone ive got an issue trying to connect from internet, through NAT to asterisk. |
| 03:25.31 | p3nguin | looks around to see if anyone saw him |
| 03:25.37 | nobodyshome | getting connection refused. |
| 03:25.58 | nobodyshome | ports are forwarded. 5060-5080 & 10001 - 20000 DUP |
| 03:26.00 | nobodyshome | UDP |
| 03:26.28 | p3nguin | You don't need 5061-5080, and don't forget 10000. |
| 03:27.04 | nobodyshome | ive got the firewall completely disabled right now, still nothing. ill try 10000 now. |
| 03:27.42 | p3nguin | You need UDP 5060 and whatever UDP range is specified in rtp.conf |
| 03:27.59 | p3nguin | It they need to be forwarded to the Asterisk computer. |
| 03:28.09 | nobodyshome | whats default? |
| 03:28.18 | nobodyshome | for port in rtp.conf? |
| 03:28.20 | p3nguin | It's typically 10000-20000. |
| 03:28.23 | nobodyshome | k |
| 03:28.39 | nobodyshome | port scanning is telling me their all blocked. |
| 03:28.50 | p3nguin | You're probably doing a TCP scan. |
| 03:29.08 | nobodyshome | well ive actually got both tcp& udp ranges forwarded |
| 03:29.17 | p3nguin | You don't need TCP. |
| 03:29.25 | p3nguin | But even if you forward them, they'll do nothing. |
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| 03:34.40 | *** join/#asterisk diijiib (~nobodysho@bas10-kitchener06-1279681960.dsl.bell.ca) |
| 03:34.52 | diijiib | k im back |
| 03:35.01 | p3nguin | As I was saying... |
| 03:35.05 | p3nguin | Even if you forward TCP ports, they'll do nothing because asterisk isn't using TCP. |
| 03:35.26 | diijiib | ok ive changed the ranges in dd-wrt to only udp |
| 03:35.35 | diijiib | and i realise that p3nguin |
| 03:36.07 | diijiib | its still saying the ports closed thouhg.. hmmm |
| 03:36.19 | p3nguin | You're still doing a TCP scan. |
| 03:36.29 | diijiib | no way jose' |
| 03:36.31 | p3nguin | So stop scanning and use something that uses asterisk properly. |
| 03:36.34 | p3nguin | like a phone. |
| 03:36.44 | diijiib | ive got a buddy in toronto trying to connect |
| 03:36.48 | diijiib | on his android |
| 03:36.57 | p3nguin | sip set debug on |
| 03:37.03 | diijiib | verbose 4 |
| 03:37.06 | diijiib | or debug? |
| 03:37.12 | p3nguin | sip set debug on <------ this |
| 03:38.00 | diijiib | debug 4? on didnt work |
| 03:38.05 | p3nguin | sip set debug on <------ this! |
| 03:38.13 | diijiib | ohh sip k |
| 03:38.25 | p3nguin | Am I speaking a different language tonight? |
| 03:38.29 | diijiib | check |
| 03:38.34 | diijiib | no im just dense |
| 03:38.39 | p3nguin | Fair enough. |
| 03:39.40 | p3nguin | SOME mobile clients do use TCP. If you have one of those, you have to make asterisk use TCP as well as UDP, plus forward TCP 5060. |
| 03:39.55 | p3nguin | I don't use any of them, so I don't know which ones try to use TCP. |
| 03:40.08 | diijiib | whatever the sip client the android comes with |
| 03:40.12 | diijiib | not sure what its called |
| 03:40.13 | p3nguin | no idea |
| 03:40.19 | p3nguin | sipdroid? |
| 03:40.38 | p3nguin | Do you have tcpdump on your gateway? |
| 03:41.46 | diijiib | no. |
| 03:42.12 | diijiib | howcome im seeing addresses from other subnets in the sip debug? |
| 03:42.18 | singler | after upgrade to asterisk 1.8.5 I got "chan_ooh323.c: Unacceptable ip x.x.x.x" error on incoming h323 calls, I found some info about changed peer matching. Is there some sort of insecure= option in ooh323.conf? |
| 03:42.35 | p3nguin | You're getting traffic from devices that aren't on your LAN, apparently. |
| 03:43.04 | p3nguin | remote phones, an ITSP, et cetera |
| 03:44.30 | diijiib | why are they comming to me? related to that 's' extension issue? |
| 03:44.43 | diijiib | their in 192.1682.x subnet. i use 10.0.0.x |
| 03:44.53 | p3nguin | NAT problem |
| 03:45.10 | p3nguin | Where are all of your phones? |
| 03:45.13 | diijiib | nat problem the 's' thing or the address issue |
| 03:45.18 | diijiib | behind my NAT |
| 03:45.26 | diijiib | besides the one my buddy is trying to connect |
| 03:45.40 | p3nguin | Is he on a 192.x address? |
| 03:46.52 | diijiib | no |
| 03:46.59 | diijiib | he should be comming from 24.x.x.x |
| 03:47.13 | p3nguin | Is he using wireless on his LAN? |
| 03:47.35 | p3nguin | And he has a 192.168.x.x address on his phone? |
| 03:48.56 | bougyman | t49 |
| 03:49.20 | diijiib | waiting for resonce from him |
| 03:49.41 | p3nguin | If that's the case, it's possible that you haven't configured properly for NAT on Asterisk. |
| 03:49.54 | *** join/#asterisk syntaxx (~patvan@unaffiliated/syntaxx) |
| 03:50.37 | syntaxx | hi can anyone suggest a softphone client that supports windows and linux very well? thanks! |
| 03:50.49 | diijiib | x-lite? |
| 03:50.55 | diijiib | is it on linux? |
| 03:51.07 | p3nguin | Maybe zoiper. |
| 03:52.39 | diijiib | reading: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:52.41 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 03:54.59 | diijiib | nothing i see in debug refers to the extensions he is using |
| 03:55.03 | diijiib | '300' |
| 03:55.16 | p3nguin | Of course it doesn't. He isn't using any extension. |
| 03:55.38 | diijiib | im going to slap myself stupid |
| 03:55.39 | p3nguin | Extensions are only used when a call is made. You indicated all he was doing was trying to "connect." |
| 03:55.49 | diijiib | yes |
| 03:55.56 | diijiib | ok the username |
| 03:55.56 | p3nguin | No extensions involved in that. |
| 03:55.59 | diijiib | ok |
| 03:56.09 | diijiib | i dont know the lexicon |
| 03:57.44 | diijiib | k p3nguin he is using 192.168.2.x |
| 03:58.01 | p3nguin | And that's where it came from in the sip debug. |
| 03:58.10 | p3nguin | Correctly configure for NAT. |
| 03:58.37 | p3nguin | ~sipnat |
| 03:58.37 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 03:59.22 | p3nguin | For newer versions, canreinvite is called directmedia, and externip is called externaddr. |
| 04:00.48 | p3nguin | Your asterisk is not behind a NAT if it is running on the NAT server, so you'd probably use nat=no, directmedia=no, and define the localnet of any rfc1918 subnet that will be using your asterisk. |
| 04:01.10 | p3nguin | With asterisk on the NAT server, I don't know if you need to set externhost/externaddr or not. |
| 04:01.23 | WIMPy | no |
| 04:01.44 | diijiib | asterisk is behind the nat. |
| 04:01.55 | diijiib | not the nat itself |
| 04:02.07 | p3nguin | I thought you said you were running it on dd-wrt. |
| 04:02.56 | diijiib | ok (internet) -> ddwrt NAT on wrt54g -> openwrt54g * server |
| 04:03.13 | *** join/#asterisk X-Rob_ (~Rob@eth2083.qld.adsl.internode.on.net) |
| 04:04.43 | p3nguin | Oh. Then you must set nat=yes, directmedia=no, externhost or externaddr, and the localnet settings. |
| 04:04.49 | p3nguin | all in [general] |
| 04:04.53 | p3nguin | (sip.conf) |
| 04:05.24 | p3nguin | And you also need to set nat=yes in the peer definition for your friend's phone, because he, too, is behind a NAT. |
| 04:05.55 | diijiib | externhost is defined how? |
| 04:06.23 | p3nguin | If you have a host name that is resolvable on the internet, you use externhost=yourhostname.com or whatever. |
| 04:06.38 | diijiib | no IP? |
| 04:06.43 | p3nguin | If you have a static IP address, you can use externaddr=IPaddressthatdoesntchangeever |
| 04:06.53 | p3nguin | one or the other, not both. |
| 04:08.33 | diijiib | what if its a dynamic address? |
| 04:08.42 | diijiib | the root domain of his FQDN? |
| 04:09.40 | p3nguin | That question doesn't seem to make sense to me. |
| 04:09.53 | diijiib | lol |
| 04:09.57 | diijiib | we poor people |
| 04:10.12 | p3nguin | You are configuring asterisk on your LAN. We don't care about anyone else's IP addresses or hostnames. |
| 04:10.15 | diijiib | have these addresses known as dynamic IPs |
| 04:10.44 | diijiib | then what about externaddr & externhost? |
| 04:11.03 | diijiib | ill look it up |
| 04:11.08 | p3nguin | YOUR public IP address or YOUR public host name that has a resolvable IP address. |
| 04:11.24 | p3nguin | They go in your general section of sip.conf |
| 04:12.14 | p3nguin | Asterisk 1.4 sip.conf working example: http://pastebin.com/tER2jGnY |
| 04:12.22 | diijiib | so if i have a DDNS i use that in externhost |
| 04:12.27 | p3nguin | correct |
| 04:12.37 | diijiib | ok then. |
| 04:13.04 | p3nguin | If you use externhost with your dynamic host name, don't use externaddr/externip. |
| 04:14.28 | diijiib | when i do sip show peers. i see the NAT column and under buddies 300 peer it shows NAT = N |
| 04:14.38 | diijiib | althought in sip.conf is set to YES |
| 04:14.53 | diijiib | that because he hasnt connected? |
| 04:15.09 | p3nguin | Most people only use externhost when they have a dynamic dns name, but if you have a static address and a good dns hname, you can use that too. |
| 04:15.27 | p3nguin | In the NAT column, N means NAT is enabled. Blank means not configured. |
| 04:15.35 | diijiib | ahh k |
| 04:15.47 | diijiib | well im using ddns so should be in good shape now |
| 04:16.24 | p3nguin | You have localnet values entered in your general section? |
| 04:16.53 | diijiib | yes |
| 04:17.07 | diijiib | and remote under his extension |
| 04:17.07 | p3nguin | You'll need to at least enter your own localnet there. Maybe even his localnet as well if you still have trouble. |
| 04:17.15 | p3nguin | what? |
| 04:17.24 | p3nguin | Again, your words don't make sense to me. |
| 04:17.25 | diijiib | localnet being? |
| 04:17.37 | diijiib | lol |
| 04:17.48 | p3nguin | localnet=10.0.0.0/255.0.0.0 <--- example |
| 04:17.52 | diijiib | under his user context? |
| 04:18.04 | p3nguin | There is no localnet parameter for peer entries. |
| 04:18.06 | diijiib | thats my subnet |
| 04:18.12 | p3nguin | It is only valid in the general section. |
| 04:18.18 | diijiib | what about his? |
| 04:18.22 | diijiib | also in general? |
| 04:18.33 | p3nguin | (2317.07) <p3nguin> You'll need to at least enter your own localnet there. Maybe even his localnet as well if you still have trouble. |
| 04:18.57 | p3nguin | ^this |
| 04:19.09 | diijiib | ok so wont they conflict? |
| 04:19.17 | diijiib | if mines 10 and his 192? |
| 04:19.43 | p3nguin | The localnet parameter tells asterisk which rfc1918 addresses to do NAT fix-up on. |
| 04:19.54 | p3nguin | or which ones to not do it on, rather. |
| 04:20.01 | diijiib | or 10.0.0.0/255.0.0.0,192.168.2.0/255.255.255.0 |
| 04:20.04 | p3nguin | no |
| 04:20.08 | p3nguin | Just do what I said. |
| 04:20.24 | p3nguin | Stop trying to guess that I might mean something else. |
| 04:20.34 | diijiib | so two localnet entries? |
| 04:20.39 | p3nguin | Not yet. |
| 04:20.43 | diijiib | if trouble |
| 04:21.06 | p3nguin | If more problems occur, we might put in his to at least do some debugging. |
| 04:21.13 | p3nguin | FOr now, put only yours. |
| 04:21.25 | diijiib | k mines in net/subnet |
| 04:22.08 | p3nguin | Good. Do you have, in your general section, nat=yes, directmedia=no, externhost=yourhost.domain.com, and localnet=10.0.0.0/255.0.0.0 |
| 04:22.11 | p3nguin | ? |
| 04:22.13 | diijiib | as for the phone context, same config as internal lines plus the nat=yes? |
| 04:22.23 | diijiib | YES |
| 04:22.33 | p3nguin | Good. Move on to his peer definition for his phone. |
| 04:22.54 | p3nguin | You need to add nat=yes and directmedia=no in his peer definition. |
| 04:22.54 | diijiib | ok anything differing in there from what i had for 100 & 20? |
| 04:22.56 | diijiib | 200 |
| 04:23.08 | diijiib | ahh directmedia i didnt define... yet |
| 04:23.28 | p3nguin | That keeps asterisk in the media path so it can peform nat fix-up. |
| 04:23.30 | diijiib | done. |
| 04:23.33 | p3nguin | perform |
| 04:24.04 | p3nguin | save, run sip reload, and try his phone again. |
| 04:24.10 | diijiib | cant. |
| 04:24.13 | diijiib | he's in bed. |
| 04:24.15 | p3nguin | hahaha |
| 04:24.22 | diijiib | :D |
| 04:24.40 | p3nguin | I do the same thing. Someone takes too long, I go to bed half way through what they are doing. |
| 04:25.38 | diijiib | ahh he has to work for some media firm in toronto in the am |
| 04:25.38 | p3nguin | You should open a ticket with the ITSP and ask them why you are getting calls to extension 's' from them. |
| 04:25.47 | diijiib | his regular it job |
| 04:25.59 | diijiib | yeh i should. by phone or online though? |
| 04:26.00 | *** join/#asterisk Spanktar (~SpankyFro@c-71-198-58-106.hsd1.ca.comcast.net) |
| 04:26.10 | p3nguin | Either way. |
| 04:26.32 | p3nguin | If you use the online chat in the daytime, they'll tell you to open a ticket by email. |
| 04:26.55 | diijiib | is there a writeup on the procedure? |
| 04:27.08 | p3nguin | On sending an email? |
| 04:27.17 | p3nguin | send it to support@voip.ms |
| 04:27.41 | p3nguin | Tell them you are getting calls from them to extension 's' instead of to your phone number. You want to know why and how to correct it. |
| 04:29.06 | diijiib | rogers |
| 04:29.29 | p3nguin | It could be a misconfiguration on their side or something. I've never once seen a call from voipms from my user id to s. The calls ALWAYS come from my phone number as the username to my phone number as the extension. |
| 04:29.58 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 04:30.13 | p3nguin | Call from '3149691077' to extension 3149691077 <-- example |
| 04:30.43 | p3nguin | Where yours was Call from '123456' to extension s. |
| 04:30.53 | diijiib | should i include logs? |
| 04:30.56 | p3nguin | sure |
| 04:30.57 | diijiib | errors |
| 04:30.58 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 04:31.25 | p3nguin | Show them the lines like what I just mentioned. |
| 04:32.50 | Spanktar | good evening! I'm very new to asterisk, but I've got a very basic setup (TDM410, 4 FXS modules). Everything on the server seems functional, but I have no dialtone on the phones. |
| 04:33.47 | james_zhu | make sure the dahdi channel is there |
| 04:34.08 | james_zhu | maybe the chan_dahdi.conf is not correct |
| 04:34.17 | Spanktar | ok, checking that... |
| 04:34.20 | diijiib | done and done. |
| 04:34.34 | james_zhu | show dahdi channels? |
| 04:34.43 | p3nguin | dahdi show channels |
| 04:35.50 | james_zhu | feed power to fxs? |
| 04:36.43 | Spanktar | the power is connected to the card |
| 04:37.44 | Spanktar | dahdi show channels? I do 'sudo dadhi show channels" => dahdi command not found |
| 04:38.01 | p3nguin | of course it isn't found there. |
| 04:38.01 | james_zhu | something is wrong with dahid.conf |
| 04:38.06 | p3nguin | It's an asterisk CLI command. |
| 04:38.11 | Spanktar | kk |
| 04:38.15 | james_zhu | under asterisk ,run that |
| 04:38.59 | Spanktar | asterisk*CLI> No such command 'dadhi show channels' (type 'core show help dahdi show'... |
| 04:39.13 | p3nguin | Seems like your chan_dahdi isn't loaded. |
| 04:39.22 | Spanktar | ok |
| 04:39.26 | p3nguin | module load chan_dahdi.so |
| 04:39.31 | james_zhu | should be a problem with your chan_dahdi.conf |
| 04:39.54 | Spanktar | where could I see some sort of error output to know if there's a problem? |
| 04:40.04 | Spanktar | rechecking chan_dahdi.conf |
| 04:40.15 | james_zhu | when you boot the asterisk, the error should be there |
| 04:40.17 | p3nguin | If it's fatal, it should spew onto the CLI when you do the module load on it. |
| 04:40.20 | james_zhu | asterisk -vvvvvvvvvvgc |
| 04:40.37 | Spanktar | ah yes |
| 04:40.46 | Spanktar | some warnings and some errors |
| 04:40.48 | james_zhu | ok, the led of card is blink? |
| 04:41.01 | Spanktar | no led is solid |
| 04:41.17 | james_zhu | should be in green |
| 04:41.23 | Spanktar | blue |
| 04:41.25 | james_zhu | if the driver is ok |
| 04:41.28 | james_zhu | sorry |
| 04:41.31 | diijiib | p3nguin, u are a * * * god |
| 04:41.31 | james_zhu | blue |
| 04:41.44 | p3nguin | What'd I do? |
| 04:42.02 | Spanktar | attempt to configure channel 1 with signaliing FXO ks ignored b/c it is alredy configured to be FXO ks |
| 04:42.08 | Spanktar | and... |
| 04:42.12 | james_zhu | wrong |
| 04:42.14 | diijiib | u just are that good |
| 04:42.28 | james_zhu | if fxo, should be fxs in conf |
| 04:42.42 | Spanktar | they're FXS |
| 04:42.48 | Spanktar | 4 FXS (green) |
| 04:42.52 | james_zhu | should be fxo in conf |
| 04:43.03 | Spanktar | is |
| 04:43.12 | Spanktar | double checking |
| 04:43.17 | james_zhu | show the system.conf file? |
| 04:43.26 | james_zhu | or run demsg, show the result? |
| 04:43.39 | Spanktar | system.conf: fxoks = 1-4 |
| 04:43.42 | diijiib | dmesg |
| 04:43.57 | Spanktar | btw, thank you all so much :) |
| 04:44.12 | Spanktar | echocanceller = mg2,1-4 |
| 04:44.24 | Spanktar | i'll try to get dmesg, but I'm typing these longhand |
| 04:44.37 | Spanktar | two different machines... :( |
| 04:44.44 | Spanktar | let me try to ssh in |
| 04:44.54 | p3nguin | You need curlpaste or pastebinit (if that one still works). |
| 04:44.59 | Spanktar | yeah |
| 04:45.02 | james_zhu | make sure the driver loaded without error |
| 04:45.06 | Spanktar | I think I can get to in on the LAN |
| 04:45.23 | Spanktar | ssh 192.168.1.101 |
| 04:45.26 | Spanktar | d'oh! |
| 04:45.44 | Spanktar | in! |
| 04:45.56 | syntaxx | hi can anyone suggest a softphone client that supports video on windows and linux very well? thanks! |
| 04:46.00 | diijiib | *shakes head* at subnet |
| 04:46.20 | p3nguin | Now what happened? |
| 04:46.48 | diijiib | linphone |
| 04:46.55 | Spanktar | ha! the subnet's not my choice. rommate |
| 04:47.29 | Spanktar | http://pastebin.com/3Tg8ukpe |
| 04:47.33 | Spanktar | ^^ dmesg |
| 04:50.04 | diijiib | WARNING: at /home/spanky/dahdi-linux-complete-2.5.0+2.5.0/linux/drivers/dahdi/wctdm24xxp/base.c:1996 wctdm_fxs_hooksig+0xd6/0x1e0 [wctdm24xxp]() |
| 04:50.04 | diijiib | [ 25.226723] Hardware name: XPST550 |
| 04:50.04 | diijiib | [ 25.226731] 0 is an invalid signaling state for an FXS module. |
| 04:50.07 | diijiib | issue? |
| 04:50.10 | Spanktar | http://pastebin.com/PxvNFYW7 <- system.conf |
| 04:50.35 | james_zhu | show your chan_dahdi.conf |
| 04:50.38 | Spanktar | kk |
| 04:50.55 | james_zhu | without dialtone is because of loading chan_dahdi.conf wrongly |
| 04:51.43 | Spanktar | http://pastebin.com/AXCQmAgx <â chan_dahdi.conf |
| 04:52.00 | Spanktar | i cut off the boilerplate at the top. lemme know if you need that, too |
| 04:52.54 | Spanktar | i think i see it |
| 04:53.01 | Spanktar | signalling = fxs_ko |
| 04:53.08 | Spanktar | should be kS |
| 04:53.18 | james_zhu | =-O |
| 04:53.54 | Spanktar | how to 'reload'? |
| 04:54.08 | james_zhu | module reload chan_dahdi.so |
| 04:54.34 | james_zhu | you should see it success or fail |
| 04:54.44 | Spanktar | i have a NOTICE |
| 04:54.46 | Spanktar | one sec |
| 04:54.57 | james_zhu | some spelling is wrong, the system will show that |
| 04:55.17 | Spanktar | [Aug 8 21:55:14] NOTICE[1795]: loader.c:721 ast_module_reload: The module 'chan_dahdi.so' was not properly initialized. Before reloading the module, you must run "module load chan_dahdi.so" and fix whatever is preventing the module from being initialized. |
| 04:55.49 | james_zhu | module load chan_dahdi.so |
| 04:55.56 | Spanktar | yeah, doing... |
| 04:55.57 | james_zhu | or add modules.conf |
| 04:56.50 | Spanktar | you know what? I think I should just yank the whole [Incoming] section |
| 04:56.51 | Spanktar | b/c |
| 04:56.56 | Spanktar | [Aug 8 21:56:09] ERROR[1795]: chan_dahdi.c:16522 build_channels: Unable to register channel '' |
| 04:57.08 | Spanktar | and I have |
| 04:57.18 | Spanktar | ; FXO Modules |
| 04:57.24 | Spanktar | channel = |
| 04:57.27 | Spanktar | removing |
| 04:58.20 | Spanktar | much better |
| 04:58.30 | Spanktar | dialtone!! |
| 04:58.35 | Spanktar | w00t! |
| 04:59.05 | Spanktar | hmmm. should I worry about this? |
| 04:59.07 | Spanktar | [Aug 8 21:58:46] ERROR[1918]: cdr_csv.c:314 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied |
| 04:59.31 | Spanktar | seems like it just can't write to the logfile |
| 04:59.37 | james_zhu | chmod 777 /var/log/asterisk |
| 05:00.03 | p3nguin | That's not the right way to handle it. |
| 05:00.17 | Spanktar | +1 |
| 05:00.26 | Spanktar | i mean, that *will* solve it ;)_ |
| 05:00.30 | p3nguin | chown asterisk /var/log/asterisk ; chmod 755 /var/log/asterisk |
| 05:00.52 | Spanktar | drwxr-xr-x 2 root root 4.0K Aug 8 20:33 cdr-csv/ |
| 05:00.53 | Spanktar | drwxr-xr-x 2 root root 4.0K Aug 8 20:33 cdr-custom/ |
| 05:00.53 | Spanktar | drwxr-xr-x 2 root root 4.0K Aug 8 20:33 cel-custom/ |
| 05:00.56 | Spanktar | i'll chown |
| 05:01.19 | p3nguin | All that stuff should be owned by asterisk:asterisk and asterisk should run as uid asterisk. |
| 05:01.27 | Spanktar | yup |
| 05:01.33 | Spanktar | i think my make install was sudo |
| 05:02.06 | Spanktar | but everything seems to be running as asterisk in top |
| 05:02.10 | Spanktar | so thay's good |
| 05:02.38 | p3nguin | # namei -mo /var/log/asterisk/cdr-csv/ |
| 05:02.38 | p3nguin | f: /var/log/asterisk/cdr-csv/ drwxr-xr-x root root / drwxr-xr-x root root var drwxr-xr-x root root log drwxr-xr-x asterisk asterisk asterisk drwxr-xr-x asterisk asterisk cdr-csv |
| 05:02.46 | p3nguin | Well that broke. |
| 05:03.15 | Spanktar | yeah, that's what i have now |
| 05:03.18 | p3nguin | http://pastebin.com/hUQ8qDyQ |
| 05:03.49 | Spanktar | http://pastebin.com/QgdU12cC |
| 05:03.55 | Spanktar | no errors on the console :) |
| 05:04.04 | Spanktar | everything's working great now |
| 05:04.08 | Spanktar | thank you guys so much |
| 05:04.21 | Spanktar | I've consistently gotten great help in this channel |
| 05:04.26 | Spanktar | reminds me of #plone |
| 05:04.39 | Spanktar | very helpful bunch there too |
| 05:05.05 | Spanktar | again, thanks a lot for you help, not only did I solve the problem, I also learned a thing or two about how to find the errors |
| 05:05.14 | Spanktar | ...teach a man to fish... |
| 05:05.36 | p3nguin | Now make me some sushi! |
| 05:05.44 | diijiib | mm sushi |
| 05:05.45 | Spanktar | i can do that!! |
| 05:06.50 | diijiib | so p3nguin what gui if any on a wrt54g would u suggest for asterisk? -gui or freepbx? |
| 05:06.59 | p3nguin | none |
| 05:07.09 | diijiib | none |
| 05:07.13 | diijiib | ok then |
| 05:07.15 | diijiib | y? |
| 05:07.38 | p3nguin | GUIs just take away all the usefulness of Asterisk. |
| 05:07.57 | diijiib | yeah but im limited by my own uselessness |
| 05:08.09 | p3nguin | If you use a GUI, you'll remain that way. |
| 05:08.28 | diijiib | eye nose |
| 05:08.44 | diijiib | k then useful apps? wakup call? etc? |
| 05:08.49 | diijiib | for home use |
| 05:08.56 | p3nguin | Do you need a wakeup call? |
| 05:09.02 | diijiib | and how! |
| 05:09.07 | p3nguin | I use crontab for such things. |
| 05:09.18 | p3nguin | But that's the sysadmin in me. |
| 05:09.26 | diijiib | i hear ya |
| 05:09.54 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:10.10 | p3nguin | If you need it, there's a wakeup perl script out there. |
| 05:10.13 | diijiib | im more thinking for the GF & guests for ease of use |
| 05:10.20 | diijiib | i saw that |
| 05:10.24 | p3nguin | I set it up once, but it never worked. |
| 05:10.41 | p3nguin | I didn't need it, so I didn't ever get around to finding out the problem. |
| 05:10.49 | diijiib | k |
| 05:10.56 | p3nguin | I may have done it wrong. |
| 05:10.59 | WIMPy | I use at. |
| 05:11.03 | p3nguin | at is good, too. |
| 05:11.12 | diijiib | thats a cron job no? |
| 05:11.18 | WIMPy | But I really sould do somethign where you can delete an alarm again. |
| 05:11.22 | p3nguin | I use at on my desktop more than on a server, though. |
| 05:11.31 | p3nguin | No, at uses atd, cron uses crond. |
| 05:11.33 | diijiib | at is a windows command no? |
| 05:11.45 | diijiib | ok then i stand corrected |
| 05:11.57 | Spanktar | p3nguin, diijiib: me teaching my friend's daughter to roll sushi :) http://twitpic.com/63a2lh |
| 05:12.18 | diijiib | lol |
| 05:12.27 | WIMPy | The good thing about at is that it will take the time as within the next 24h. |
| 05:12.39 | Spanktar | 'twas delicious |
| 05:12.49 | diijiib | too appropriet Spanktar |
| 05:13.06 | Russ | Spanktar, you should make some kimbap too |
| 05:13.20 | Spanktar | i make a decent Ramen |
| 05:13.20 | p3nguin | at is much quicker to enter and more efficient for single jobs, too. |
| 05:13.23 | Spanktar | , too |
| 05:13.44 | p3nguin | No editing of config files (crontab). |
| 05:14.23 | diijiib | but jow blow can easily do that at his desk |
| 05:14.33 | diijiib | without putty and training, and rights. |
| 05:14.45 | p3nguin | He can use 'at' as well. |
| 05:15.10 | p3nguin | at 5:00 am, ... |
| 05:15.37 | WIMPy | Or your dialplan can do it. |
| 05:15.54 | p3nguin | one-shot, then it's done. If you use crontab, you have to go back later to delete the line, even if you've configured it to only run once. |
| 05:16.00 | diijiib | whats a syntax you would use in full to call my phone at 5am |
| 05:17.33 | p3nguin | at 5:00 am, asterisk -rx 'originate SIP/100 application playback wakeup' |
| 05:17.51 | p3nguin | where SIP/100 is your phone, and wakeup is a valid sound file. |
| 05:18.20 | p3nguin | after the time, you have to press enter. Then you enter the command. |
| 05:18.35 | p3nguin | I used a comma to indicate where I should have pressed enter. |
| 05:19.55 | p3nguin | I would probably not use application playback wakeup, though; I would use extension wakeup@misc, where that extension does things like wait, play sounds files, repeat. |
| 05:20.21 | p3nguin | at 5:00 am, asterisk -rx 'originate SIP/100 extension wakeup@misc' |
| 05:20.51 | p3nguin | Make sure you have atd running. |
| 05:20.56 | diijiib | whats the x in -rx? |
| 05:21.01 | WIMPy | Or MusicOnHold(wake)? |
| 05:21.03 | p3nguin | execute a command |
| 05:21.10 | diijiib | ah |
| 05:21.15 | p3nguin | sure, moh would be perfect for that. |
| 05:21.40 | diijiib | i need to get recording working |
| 05:21.45 | diijiib | thats one thing i NEED |
| 05:21.45 | p3nguin | originate SIP/100 application musiconhold wake |
| 05:21.52 | p3nguin | Recording? |
| 05:22.01 | p3nguin | On the automixmon thing? |
| 05:22.07 | diijiib | ya either automixmon or automon or something like that |
| 05:22.20 | p3nguin | Try automon. That one works for me. |
| 05:22.23 | p3nguin | *1 |
| 05:22.28 | p3nguin | recording starts. |
| 05:22.42 | diijiib | where is the output location and is it one or two files for the streams? |
| 05:23.05 | p3nguin | automon will record two files, but you can make it merge them after it is done. |
| 05:23.21 | p3nguin | The normal output location is /var/spool/asterisk/monitor/ |
| 05:24.18 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 05:24.42 | diijiib | the call? |
| 05:25.59 | diijiib | oh i have a problem i couldnt resolve earlier that i forgot about |
| 05:26.02 | diijiib | voicemail time. |
| 05:26.17 | diijiib | openwrt reports correct time. |
| 05:26.33 | diijiib | i have tz=eastern set in voicemail.conf for my mailbox |
| 05:26.37 | diijiib | what do? |
| 05:26.40 | *** join/#asterisk djuhl30 (~quassel@121.135.82.142) |
| 05:27.05 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 05:27.11 | djuhl30 | I have a sip account with flowroute. |
| 05:28.03 | djuhl30 | I am trying to get it to go to my local asterisks server. Does the answer application only answer or can I expect it to tranfer it to my softphone? |
| 05:28.18 | WIMPy | I don;t think "eastern" is a valid TZ. |
| 05:28.25 | djuhl30 | I am trying to figure out what dialpla code I'll actually need |
| 05:28.43 | p3nguin | djuhl30: Do not Answer(), just use Dial(). No transferring needs to be done. |
| 05:28.57 | djuhl30 | hmmm |
| 05:29.06 | p3nguin | djuhl30: Answer() only answers the line and makes it go to state "Up." |
| 05:29.12 | djuhl30 | ok |
| 05:29.17 | p3nguin | You don't want that to happen until you pick up the phone. |
| 05:29.26 | djuhl30 | give me a sec to try to attempt to get a guess |
| 05:29.30 | diijiib | djuhl30, let me have ssh access to your box and ill fix you right up |
| 05:29.35 | p3nguin | Unless, of course, you do... because you're doing other stuff before you pick up the phone. |
| 05:29.40 | diijiib | i think i could handle this one p3nguin |
| 05:29.48 | p3nguin | maybe |
| 05:29.51 | diijiib | ;) |
| 05:32.08 | diijiib | djuhl30, pastebin.com your sip.conf & extensions.conf files |
| 05:32.50 | djuhl30 | So exten => 12012156850,1,Answer() should be changed to exten => 12012156850,1,Dial(SIP/12012156850,30) |
| 05:32.55 | djuhl30 | diijiib: give me a sec |
| 05:33.29 | p3nguin | If the extension you get calls to is 12012156850, and you have a SIP phone called SIP/12012156850, yes. |
| 05:34.27 | djuhl30 | http://pastebin.com/8D0BiQd4 |
| 05:35.08 | djuhl30 | p3nguin: I'll give it a try. My did is 12012156850 so I'd thought that would be from not to. |
| 05:35.54 | p3nguin | http://pastebin.com/aSgFL4BB <-- your paste, corrected |
| 05:36.05 | djuhl30 | http://pastebin.com/SvqzCs2q |
| 05:36.22 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 05:37.01 | p3nguin | Don't forget to change your password now that you've shown everyone. |
| 05:37.44 | p3nguin | I would not have a SIP phone configured with my DID number as the peer's name. |
| 05:37.56 | p3nguin | It will just cause confusion later. |
| 05:38.41 | p3nguin | Also curious why you need to use the nonstandard port 5061 for that phone. |
| 05:39.09 | djuhl30 | I was taking advice from LinuxFormat Magazine |
| 05:39.37 | djuhl30 | p3nguin: I'll change it. I just want to call my softphone |
| 05:39.46 | djuhl30 | This stuff gets confusing |
| 05:40.11 | p3nguin | I'd change it to something that reflects what the phone is, like [myphone]. |
| 05:40.29 | diijiib | lol did he not hide his password |
| 05:41.24 | djuhl30 | diijiib: Yeah your right |
| 05:41.32 | djuhl30 | I've been working on this for weeks |
| 05:41.38 | djuhl30 | So I got a little complacent. |
| 05:41.44 | djuhl30 | Now I need to change it. |
| 05:41.46 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-fwavhnitzlgaeczl) |
| 05:41.54 | djuhl30 | Cool I can receive a phone call now. |
| 05:42.39 | djuhl30 | Now I can work on small things. Like getting my did number to show up in my parents caller id. |
| 05:42.52 | djuhl30 | I haven't even looked at voicemail yet. |
| 05:43.03 | p3nguin | That's easy. Just set teh caller ID number before the Dial() on outgoing. |
| 05:43.22 | p3nguin | Set(CALLERID(num)=your#here) |
| 05:44.05 | p3nguin | I'd love to help you get it all done, but it's late and I won't be awake much longer. |
| 05:44.08 | djuhl30 | Damn I need to get a new password for flowroute too so no one mooches calls |
| 05:44.16 | p3nguin | That's what I was saying! |
| 05:44.41 | djuhl30 | p3nguin: I know. I have a one tract mind and little patience |
| 05:44.43 | p3nguin | It should be easy to change the SIP password. |
| 05:44.51 | ChannelZ | make it 'password' |
| 05:45.00 | ChannelZ | nobody ever guesses that one |
| 05:45.03 | p3nguin | It's been a while since I configured my flowroute account, so I don't recall where to do it. |
| 05:45.04 | djuhl30 | ChannelZ: very funny lol |
| 05:45.20 | djuhl30 | I know how. |
| 05:45.24 | p3nguin | okay |
| 05:45.33 | djuhl30 | I am not as dumb as I look. lol |
| 05:45.38 | p3nguin | I use super-duper passwords, so I never change them. |
| 05:46.58 | p3nguin | And I keep low balances, so I don't care if you crack it anyway. |
| 05:47.38 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 05:47.41 | schmidts | good morning |
| 05:48.12 | djuhl30 | password changed |
| 05:49.23 | p3nguin | I'm off for the night, so look me up in 10 hours if no one helps you while I'm gone. (I'm sure someone will help you; they're good like that.) |
| 05:51.46 | schmidts | gn8 p3nguin |
| 05:56.06 | djuhl30 | I am not getting the callerid application syntax right |
| 05:56.24 | schmidts | djuhl30 sorry missed the beginning ;) whats your problem |
| 05:56.25 | djuhl30 | Hard to find in the book |
| 05:56.39 | djuhl30 | I can receive phone calls now |
| 05:56.51 | schmidts | thats a good point to start ;) |
| 05:56.54 | p3nguin | Set(CALLERID(num)=your#here) |
| 05:57.07 | djuhl30 | Just trying to get it so when I call home they see my did number in their phone |
| 05:57.24 | djuhl30 | num is extenstion or did number |
| 05:57.46 | djuhl30 | 'Unknown directive: Set(CALLERID(num) |
| 05:57.53 | p3nguin | Put your phone number where I said your#here. |
| 05:58.02 | schmidts | have you tried CALLERID(number) ? |
| 05:58.14 | p3nguin | It's CALLERID(num) like I said twice. |
| 05:58.33 | djuhl30 | I see no space |
| 05:58.38 | djuhl30 | So I pasted it all in |
| 05:58.40 | p3nguin | Someone show him how to use Set() in dial plan. |
| 05:58.52 | p3nguin | I'm on my way out, or I'd do it for you. |
| 05:59.40 | schmidts | p3nguin roger ;) |
| 05:59.44 | djuhl30 | I got it |
| 05:59.46 | djuhl30 | it works |
| 06:03.12 | schmidts | how did you do it djuhl30? |
| 06:03.44 | djuhl30 | I didn't I just had it set to my number |
| 06:03.54 | *** join/#asterisk james_zhu (~Administr@183.16.215.92) |
| 06:03.58 | djuhl30 | If I can set a name that would be great. |
| 06:04.07 | djuhl30 | But they know it will be from me. |
| 06:04.45 | schmidts | djuhl30 just use Set(CALLERID(name)=djuhl30) ;) |
| 06:05.22 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:06.37 | schmidts | but the name field only works for sip, iax but not over isdn or pots |
| 06:07.06 | djuhl30 | It's all sip |
| 06:07.11 | WIMPy | It can work on isdn as well. |
| 06:07.18 | djuhl30 | but all I can see is my number |
| 06:07.36 | djuhl30 | I guess I'll try to call my sister if it isn't too late in the USA |
| 06:07.41 | djuhl30 | See what she sees |
| 06:08.08 | djuhl30 | Now all I need to hunt down is a South Korean did. Google gives me dead ends |
| 06:09.17 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 06:11.32 | djuhl30 | That and a cheap voip phone so I don't need to use my computer anymore |
| 06:12.40 | schmidts | djuhl30 ;) |
| 06:13.06 | djuhl30 | I need to think of some good names to call my phones too. |
| 06:13.19 | djuhl30 | This can get confusing in a heart beat. |
| 06:14.07 | djuhl30 | I can't open up my server to the internet either, |
| 06:14.43 | djuhl30 | I need to get incontact with the land lord so he can open me up to the ISP provider. All my router is cabled into is a switch. |
| 06:15.07 | djuhl30 | I don't pay for internet. |
| 06:15.19 | djuhl30 | Well I do indirectly it comes with the rent |
| 06:17.10 | schmidts | :D sounds cool |
| 06:17.20 | schmidts | atleast its the same for me, but i am working for an isp ;) |
| 06:18.03 | djuhl30 | I am in the Army |
| 06:18.15 | djuhl30 | This is a hobby that drives my wife nuts |
| 06:18.24 | schmidts | what the army? |
| 06:18.29 | djuhl30 | USA |
| 06:18.42 | djuhl30 | You know soldier. |
| 06:18.49 | djuhl30 | Go to Iraq and have some fun |
| 06:19.00 | djuhl30 | Blah Blah |
| 06:19.15 | schmidts | i see :D |
| 06:19.24 | djuhl30 | But I am slowly understanding this |
| 06:19.32 | djuhl30 | Read a paragraph or 2 |
| 06:19.41 | djuhl30 | play until I get confused. |
| 06:20.04 | schmidts | the mostly the best way to work with the book |
| 06:20.47 | djuhl30 | Do I want voicemail first or conference calling? |
| 06:21.17 | djuhl30 | I better backup my configurations I got now, so I have something to fall back on if I bork things up. |
| 06:21.30 | *** join/#asterisk Spanktar (~SpankyFro@c-71-198-58-106.hsd1.ca.comcast.net) |
| 06:22.59 | *** join/#asterisk Spanktar (~SpankyFro@c-71-198-58-106.hsd1.ca.comcast.net) |
| 06:31.33 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 06:36.32 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 06:38.31 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 06:43.43 | *** join/#asterisk kaii (~kh@ciphron.de) |
| 06:43.49 | kleszcz | format_wav.c:233 update_header: Unable to find our position |
| 06:44.15 | ChannelZ | GPS is broken |
| 06:45.13 | kaii | maybe someone didnt pay the bill ;) |
| 06:46.13 | djuhl30 | I called my wife's phone and she can barely hear me. Can I do any configurations on asterisks to make me loader? |
| 06:46.20 | djuhl30 | I just called her cellphone |
| 06:46.26 | ChannelZ | not unless you're on analog |
| 06:46.42 | djuhl30 | hmm |
| 06:46.51 | ChannelZ | What were you calling her from? |
| 06:47.01 | djuhl30 | my softphone |
| 06:47.11 | djuhl30 | But it is an American did |
| 06:47.24 | ChannelZ | Well that's more to do with your mic setup then |
| 06:47.30 | ChannelZ | What platform? |
| 06:47.45 | djuhl30 | My mic is fine |
| 06:47.51 | djuhl30 | I can hear her fine |
| 06:48.00 | djuhl30 | she can't hear me when I call her cellphone number |
| 06:48.01 | ChannelZ | That has nothing to do with your mic |
| 06:48.07 | ChannelZ | (you hearing her) |
| 06:48.36 | djuhl30 | She can't hear me and the incall volume is set up all the way |
| 06:48.50 | ChannelZ | Anyways it's all digital so dollars-to-doughnuts your level going into the softphone is low |
| 06:49.17 | djuhl30 | So if I get a VOIP phone the problem will go away? |
| 06:49.26 | ChannelZ | no |
| 06:49.32 | ChannelZ | well, yes and maybe no |
| 06:49.33 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 06:49.42 | ChannelZ | Your softphone is a voip phone |
| 06:49.44 | kaldemar | func VOLUME can be used to increase volume on channels despite technology. |
| 06:49.58 | ChannelZ | but your mic is going into the computer somehow.. Either the input in the software mixer is low, or the mic just sucks, etc. |
| 06:50.01 | djuhl30 | It's all the way up and she still can't hear me |
| 06:50.22 | djuhl30 | But I can call the USA and they hear me just fine |
| 06:50.59 | ChannelZ | She has a US cell number? |
| 06:52.00 | Gugge | djuhl30: setup an echo test on asterisk and check |
| 06:52.26 | ChannelZ | or rather Playback() one of the system sounds, have her call in and see if that's quiet or not |
| 06:52.44 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 06:53.46 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 06:53.49 | Gugge | that would work to, but with the echo test he should easily hear if his voice is as loud as the prompts |
| 06:53.59 | Gugge | and it wouldnt require outside help :) |
| 06:54.09 | ChannelZ | eh? |
| 06:55.06 | djuhl30 | I had a echo test set up once. |
| 06:55.10 | ChannelZ | Playing back a system sound which should be at a good level will pretty much prove whether it's the mic/softphone or not |
| 06:55.18 | Gugge | yes |
| 06:55.41 | ChannelZ | Echo just adds another variable, at least for narrowing down this particular problem |
| 06:55.56 | Gugge | and doing that with the echo test he can hear the difference on his own voice and the system prompts himself, without oudside help from the wife |
| 06:56.46 | djuhl30 | Cool I'll give it a try. |
| 06:57.01 | djuhl30 | You think It could just be the cellphone provider? |
| 06:57.26 | ChannelZ | It could be yes |
| 06:58.30 | *** join/#asterisk den512 (~dradon@carbon.gonicus.de) |
| 07:00.50 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 07:01.46 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 07:03.19 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:08.11 | merlin8282 | What is the CONSOLE=Console/dsp global variable used for ? |
| 07:09.35 | merlin8282 | in the sample file it's used only here: exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)})) |
| 07:09.47 | merlin8282 | but I don't understand what it exactly does. |
| 07:10.39 | kaldemar | just as an alias for the Console/dsp channel (i.e. using a sound card as a channel), if you use it. |
| 07:10.55 | merlin8282 | Aah ! Ok, thanks kaldemar ! |
| 07:19.26 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:19a3:e253:a853:8ac8) |
| 07:30.06 | *** join/#asterisk gnude (~gnude@p578bb787.dip0.t-ipconnect.de) |
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| 07:44.30 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 07:46.30 | jacc0 | hi all! good morning |
| 07:56.47 | aberrios | 'Sup |
| 07:58.01 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 07:58.29 | gnude | hello |
| 07:58.33 | gnude | good morning |
| 08:00.59 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 08:02.16 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 08:08.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 08:16.09 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:22.10 | jacc0 | I filed a bug report about a memory leak: https://issues.asterisk.org/jira/browse/ASTERISK-18247?focusedCommentId=180864#comment-180864 |
| 08:28.06 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:28.18 | krion | schmidts: thanks again for the tips, work great :) |
| 08:28.45 | jacc0 | :) |
| 08:30.47 | jacc0 | can someone clearify to me what timing modules there are and what I need to disable in menuselect to make sure only dahdi is used ? |
| 08:31.21 | *** join/#asterisk james_zhu (~Administr@183.16.215.92) |
| 08:33.42 | jacc0 | or point me to some info page |
| 08:34.26 | jacc0 | I'm talking asterisk 1.8.5 |
| 08:35.55 | jacc0 | found something on the review board |
| 08:37.03 | schmidts | krion |
| 08:37.22 | schmidts | your welcome ;) |
| 08:41.30 | *** join/#asterisk dokma (~kvirc@dh207-36-131.xnet.hr) |
| 08:42.21 | dokma | is asterisk the way to go if I want to use my external 56K USR modem to make ordinary phone calls through my Debian box using a headset and perhaps record the conversation ?? |
| 08:43.34 | Chainsaw | dokma: You need a telephony adapter. |
| 08:43.37 | Chainsaw | dokma: A modem won't do. |
| 08:43.58 | dokma | what is a telephony adapter? |
| 08:44.07 | dokma | can I read somewhere about that? |
| 08:44.30 | *** join/#asterisk Luyt (~Luyt@gandhi.xs4all.nl) |
| 08:44.36 | Chainsaw | dokma: Sure. http://www.digium.com/en/products/hardware/analog |
| 08:45.00 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85f2.bcn.adamo.es) |
| 08:45.31 | Chainsaw | dokma: Resist the urge to buy cheap & nasty X100 cards. Just stick a single module on a 4-port if you only want to use 1 line. (But really, get 1 FXO & 1 FXS so your computer can sit in the middle) |
| 08:46.04 | dokma | I don't know what FXOs or FXSs are either |
| 08:46.19 | Luyt | Would Asterisk be able to signal an application (via an UDP packet, for example) whenever a VOIP call, possible from an external source, is realized? Or does someone know of a program which sniffs SIP traffic and detect such an event? |
| 08:46.24 | dokma | it sounds like a major overkill to use all that just to use my PC as a ordinary phone |
| 08:47.59 | Chainsaw | dokma: You can leave it to someone else to purchase the appropriate hardware. |
| 08:48.06 | Chainsaw | dokma: And get an account with a VoIP telephony provider. |
| 08:48.44 | dokma | still sounds like an overkill, is it trully not possible to just use an ordinary 56K external modem |
| 08:48.45 | Chainsaw | dokma: But interfacing with the telephone network for voice requires hardware. Not a modem, because that is for data. |
| 08:48.54 | dokma | all I need is the ability to make normal phone calls |
| 08:48.58 | dokma | nothing fancy at all |
| 08:49.05 | Chainsaw | dokma: Yes. And all a modem will do is *SCREECHY MODEM NOISE HERE*. |
| 08:49.15 | dokma | ahhh.... I see |
| 08:49.26 | Chainsaw | dokma: They are great at data. Decent at fax too. |
| 08:49.31 | Chainsaw | dokma: Voice? Not so much. |
| 08:50.08 | dokma | is there some PCI card that I could plug into the PC and use as that like an ordinary phone on linux? |
| 08:50.18 | *** join/#asterisk gravin (~gravin@113.210.127.66) |
| 08:50.29 | dokma | just plug my phone line into it and make calls via my headset |
| 08:51.51 | Chainsaw | dokma: You might be able to persuade your modem to dial. But then it won't record. |
| 08:51.56 | Chainsaw | dokma: Also, Asterisk is not suitable for that. |
| 08:52.05 | dokma | I see |
| 08:52.16 | dokma | thanks a bunch for your info |
| 08:52.19 | dokma | I appreciate it |
| 08:54.07 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:54.57 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 08:55.09 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 08:58.14 | DrDigital | how would i find out the username and password for manager.conf |
| 08:59.30 | jacc0 | @DrDigital: ? in manager.conf |
| 09:00.08 | jacc0 | try "sudo su"before edeting manager.conf |
| 09:00.23 | DrDigital | I'm editing it fine? so its whatever i want it to be? |
| 09:00.35 | jacc0 | yes |
| 09:01.05 | DrDigital | i thought i had to match it to something somewhere in like mysql |
| 09:01.13 | DrDigital | I'm getting errors is why i asked |
| 09:01.39 | DrDigital | The module was unable to connect to the Asterisk manager. Make sure Asterisk is running and your manager.conf settings are proper. |
| 09:01.46 | DrDigital | its running, calls work |
| 09:01.59 | DrDigital | trying to change the MOH and cant |
| 09:04.08 | jacc0 | what module? |
| 09:04.32 | DrDigital | http://pastebin.com/7TAu5j3Z |
| 09:04.55 | DrDigital | its freepbx, I'm clicking in tools asterisk info |
| 09:05.12 | DrDigital | but i can't apply changes i get similar error |
| 09:05.41 | DrDigital | http://pastebin.com/CW1Hc3Ms |
| 09:06.10 | DrDigital | its after doing a security patch to get ride of a backdoor login |
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| 09:08.19 | *** part/#asterisk dokma (~kvirc@dh207-36-131.xnet.hr) |
| 09:09.14 | kaldemar | DrDigital: #freepbx |
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| 09:26.05 | *** join/#asterisk oktay (~oktay@81.215.202.193) |
| 09:26.18 | oktay | Hello. What documentation should I look at for Conference Calls on 1.8.x ? |
| 09:27.32 | oktay | is it MeetMe or something else? |
| 09:27.49 | kaldemar | meetme of confbridge. |
| 09:28.05 | jacc0 | @kaldemar: nederlands? |
| 09:28.07 | oktay | MeetMe seems to be a proper kind of "call in at this hour" setup right? |
| 09:28.18 | oktay | I probably don't need that much |
| 09:28.33 | oktay | just to join 3+ calls together |
| 09:28.57 | oktay | I am checking out ConfBridge. Thanks kaldemar |
| 09:29.33 | kaldemar | jacc0: nee |
| 09:32.15 | oktay | kaldemar: what doy ou know about Conference vs Konference ? |
| 09:33.25 | kaldemar | konference supports video conferencing and it is a 3rd party app. that's pretty much what i know about it as opposed to confbridge. |
| 09:33.51 | oktay | ah ok. so. I thougt I had to pick one of Conference and Konference |
| 09:33.58 | oktay | but it seems they are ConfBridge alternatives |
| 09:34.10 | oktay | and I don't need video just yet |
| 09:34.12 | oktay | thanks. |
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| 09:35.00 | james_zhu | anyone has a solution with call centor+GPRS |
| 09:35.07 | james_zhu | please note me |
| 09:35.27 | james_zhu | elastix(popup)+wireless GPRS |
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| 09:43.33 | defswork | Has anyone here use bria on android and successfully set it up to connect to * ? |
| 09:47.09 | oktay | wow it's harder than i thought to find docs on ConfBridge 1.8.5 |
| 09:47.47 | Chainsaw | oktay: Yes, it's not quite a drop-in replacement of MeetMe. |
| 09:47.56 | Chainsaw | oktay: core show application ConfBridge |
| 09:47.59 | Chainsaw | oktay: That's the best you get. |
| 09:55.01 | oktay | Chainsaw: those commands it shows. They are CLI commands ? |
| 09:55.27 | Chainsaw | oktay: That will give you the parameters for ConfBridge in the dialplan. |
| 09:55.50 | oktay | oh. ok. |
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| 10:08.00 | hariom | How to use EAGI on a remote server ? I am learning to do fast agi. I invoke my agi script using AGI(agi://127.0.0.1). Now I want to try EAGI. How to do that? |
| 10:08.30 | hariom | documentation says that its send the incoming audio on stderr. How to access stderr on a remote server ? |
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| 10:12.51 | hariom | Anyone has ever trying eagi ? |
| 10:12.59 | hariom | Anyone has ever tried eagi ? |
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| 10:19.27 | kjs | hmmm is 1.8 stable ? |
| 10:19.33 | kjs | or should i stick with 1.6 ? |
| 10:20.52 | oktay | "NOTE: This application will not automatically answer the channel." what does this mean? |
| 10:25.02 | petern_ | hariom, i don't think EAGI can be done remotely |
| 10:25.07 | oktay | anybody have a confbridge config ? |
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| 10:26.14 | hariom | petern: My conceptual block is how to access stderr on a socket communication ? |
| 10:26.26 | petern_ | hariom, you don't |
| 10:27.36 | petern_ | to do EAGI remotely would require a new protocol to separate the audio and command data |
| 10:27.40 | hariom | petern_: If normal AGI is can push its stdout to socket to be read by remote app and read from socket get as stdin, then why can't AGI push stderr on socket? |
| 10:28.59 | petern_ | because then you've got 2 data streams on 1 socket, which isn't going to work |
| 10:29.51 | petern_ | alternatively you could have another socket, but then you need some arrangement to logically link them on the app side |
| 10:30.38 | hariom | petern_: Didn't get your last msg. How would having another socket will be helpful. |
| 10:31.06 | *** join/#asterisk Weazel (~Weazel-@213.8.83.6) |
| 10:31.08 | petern_ | it's not useful unless you're going to implement remote EAGI yourself |
| 10:31.26 | Weazel | hey guys i've been busting google in search of a solution to Meetme BLF status, to be able to see if the conference room is occupied via BLF button. any suggestions - i'm very noob sry in advance |
| 10:32.14 | djuhl30 | What would be some cheap, simple VOIP phones to buy? I am using Ekiga and it would be nice to have a actual phone. |
| 10:32.42 | hariom | petern_: what if stderr is just assigned to stdout when the protocol is EAGI and record command is issued. |
| 10:33.16 | hariom | petern_: any idea if somebody has done work on remote eagi ? |
| 10:33.19 | Weazel | djuhl30: try Yealink, very cheap and very powerful |
| 10:34.29 | djuhl30 | I'll google |
| 10:34.33 | petern_ | heh, cisco rip offs :D |
| 10:34.48 | djuhl30 | Just looking for suggestions... |
| 10:34.51 | aberrios | djuhl30, Could checkout the Grandstream BudgeTone. |
| 10:36.25 | kaldemar | ~gs |
| 10:36.25 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
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| 10:38.01 | oktay | google is flooded with the NEW confbrdige. which isn't even released yet |
| 10:38.07 | oktay | btu no config for the current :) |
| 10:38.48 | aberrios | lol, kaldemar. To be honest I use Polycom but having used both Grandstream and Polycom, when it comes to budget phones I'd still consider Grandstream. |
| 10:39.27 | aberrios | I don't quite understand the bad press Grandstream have. |
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| 10:44.36 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 10:47.35 | djuhl30 | Anything else? |
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| 10:51.39 | oktay | pbx.c:4071 pbx_extension_helper: No application 'Answer ' for extension (users, 7000, 1) |
| 10:51.44 | oktay | why would I get this? |
| 10:51.51 | oktay | ah. a space. |
| 10:58.07 | oktay | ah. finally. |
| 10:58.48 | Weazel | nobody has anything about hinting meetme as a BLF ? |
| 10:59.07 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:19.23 | schmidts | Weazel where is the problem? exten => 1234,hint,Meetme(xyz) |
| 11:20.01 | Weazel | schmidts: i've tried that.. doesn't work while regular extensions have BLF without any problems |
| 11:20.30 | schmidts | Weazel what do you see when you make a core show hints |
| 11:20.58 | Weazel | State:Unavailable on the meetme |
| 11:21.45 | Weazel | schmidts ^ |
| 11:22.00 | schmidts | and meetme list ? |
| 11:22.11 | schmidts | is this an active meetme room or only generated dynamic? |
| 11:22.36 | Weazel | schmidts: 7878@ext-meetme : MeetMe:7878 State:Unavailable Watchers 0 | thats the whole line i see |
| 11:22.50 | schmidts | and what do you see with "meetme list" |
| 11:23.26 | Weazel | schmidts: " 7878 0001 N/A 00:00:01 Static No " |
| 11:24.56 | schmidts | then the state unavailable isnt wrong imho ;) |
| 11:26.53 | Weazel | yea but why the watchers is set to "0" instead of at least my BLF ? |
| 11:27.37 | schmidts | what do you see with sip show subscriptions? |
| 11:29.14 | Weazel | schmidts: the meetme extensions are not listed |
| 11:31.27 | schmidts | hm ok, maybe something went wrong with the subscribe try to do a sip debug and check the subscription |
| 11:32.54 | Weazel | schmidts: you mean debug on the peer that the BLF is set on? |
| 11:33.14 | schmidts | yes |
| 11:35.06 | Weazel | schmidts: the only thing i see strange is > SIP/2.0 404 Not Found in one of the sip packets when it tries to locate the SUBSCRIBE on the server's IP, but i'm a very noob sip debugger |
| 11:39.57 | schmidts | the 404 would mean that you subscribe to a different extension then your hint is set to |
| 11:41.52 | Weazel | schmidts: the line in the conf says "exten => 7878,hint,MeetMe:7878" isn't that right? |
| 11:42.15 | schmidts | i am not sure but i guess it should be MeetMe(7878) |
| 11:43.00 | schmidts | ok no MeetMe: should be fine |
| 11:45.39 | Weazel | schmidts: bah :( |
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| 11:46.42 | kaii | Weazel: most possibly, the lookup to match the subscribe to an extension is going to the wrong context. |
| 11:47.12 | kaii | Weazel: do you have a subscribecontext set in sip.conf? in which context is the meetme hint located? |
| 11:47.31 | Weazel | kaii: checkign |
| 11:49.38 | Weazel | kaii: is it supposed to show up as 'subscribecontext' ? or something else? coz i can't find a subscribecontext |
| 11:55.43 | Weazel | YES |
| 11:55.52 | Weazel | kaii: thank so much man, figured it out |
| 11:56.17 | Weazel | schmidts: thanks to you too for trying to help much appriciated |
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| 11:58.24 | merlin8282 | Does someone know how to do a 3-way conference ? I already found http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO, but do you know another way to do it ? |
| 12:04.19 | kaldemar | merlin8282: as a normal 3-way call (done with the phone if it supports it) or with your preferred conference application. |
| 12:04.49 | merlin8282 | kaldemar: do you have an example ? I never did this, and i'm completely lost :/ |
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| 12:06.19 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 12:06.41 | MrTelephone | is there something else I should do besides putting facilityenable=yes in chan_dahdi for getting the caller name from an ATLAS550? |
| 12:09.39 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 12:11.20 | kaldemar | merlin8282: make a call, press a button on your phone, make another call, press a button on your phone. |
| 12:12.16 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 12:12.21 | merlin8282 | kaldemar: no, I mean an example of an implementation. I know how it should work, but I don't know the implementation (and the idea behind it) |
| 12:15.57 | *** part/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 12:24.19 | kaldemar | merlin8282: a phone does the job for sip. you don't need to implement anything on the asterisk side. |
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| 12:30.31 | merlin8282 | kaldemar: mmm, Ekiga doesn't seem to implement 3-way calling :/ |
| 12:31.23 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 12:31.58 | eXcAliBuR | I'm having difficulty with getting my polycom IP321 setup with provisioning. Could I get aide here or is there a better channel? |
| 12:32.26 | MrTelephone | load it up and go through the web interface to put in the login information? |
| 12:32.42 | MrTelephone | what are you having trouble with? |
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| 12:32.47 | eXcAliBuR | when I do it that way, it works... but I don't want to do that 20 times |
| 12:33.03 | eXcAliBuR | also the dialplan needs tweaking |
| 12:33.16 | MrTelephone | dialplan is in asterisk |
| 12:33.38 | eXcAliBuR | when I do my 100 extensions it only does 10 and trys to call... it won't let me do 3 digits |
| 12:33.39 | MrTelephone | I provision using http and setup APACHE with hdav support so the phones and upload and download files as needed |
| 12:34.12 | eXcAliBuR | it's the dialplan of the phone that needs tweaking |
| 12:34.57 | MrTelephone | well you have to setup tftp or http for provisioning and then read the docs on what files to put in there. There is a generic config and then each phone has a config. |
| 12:35.20 | eXcAliBuR | i have my tftp working and it grabs a bunch of files then gives error 0X20 |
| 12:35.23 | eXcAliBuR | config error |
| 12:35.25 | eXcAliBuR | :( |
| 12:35.44 | MrTelephone | where did you get your config files? |
| 12:36.05 | eXcAliBuR | I downloaded them from polycom support |
| 12:36.24 | eXcAliBuR | might not have the right ones |
| 12:36.32 | eXcAliBuR | it's all very confusing |
| 12:36.45 | MrTelephone | I know, even worse when you have different model phones out there |
| 12:37.15 | MrTelephone | where do you see the 0x20 error? |
| 12:37.26 | eXcAliBuR | on the phone |
| 12:37.35 | eXcAliBuR | after it loads or trys to load sip.ld |
| 12:37.37 | MrTelephone | does it show "loading sip image"? |
| 12:37.39 | MrTelephone | oh |
| 12:38.32 | *** join/#asterisk coppice (~chatzilla@116.92.39.71) |
| 12:38.36 | MrTelephone | try and download the sip.ld version that is already on your phone? |
| 12:39.15 | MrTelephone | don't try and go with the latest version at first. certain bootroms work with certain sip releases too |
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| 12:41.01 | eXcAliBuR | i'll try fresh |
| 12:41.21 | eXcAliBuR | but the renaming of the files with mac addresses |
| 12:41.22 | MrTelephone | will the phone still load without the sip.ld? |
| 12:41.26 | eXcAliBuR | do i include the [ ] |
| 12:42.18 | MrTelephone | the global config file used to be phone1.cfg |
| 12:42.38 | eXcAliBuR | if I do it with the defaults it will work |
| 12:42.47 | eXcAliBuR | but i don't want defaults |
| 12:43.12 | MrTelephone | something like the dialplan you want to take affect on all phones then you change it in the global config file |
| 12:43.36 | MrTelephone | this is in my sip.cfg file |
| 12:43.37 | MrTelephone | <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3|3|3|3|3|3"/> |
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| 12:44.23 | MrTelephone | so if i change that it should take affect on all phones I would assume. I never tried. |
| 12:45.15 | MrTelephone | And when you make changes the web interface it saves the files back to your provisioning server as <MACADDRESS>-phone.cfg |
| 12:45.29 | eXcAliBuR | Really? |
| 12:45.37 | eXcAliBuR | hmmm |
| 12:45.45 | MrTelephone | if you allow write permissions in your tftp folder |
| 12:45.58 | eXcAliBuR | i'm gonna try it and see |
| 12:46.05 | eXcAliBuR | I have to jump to another network |
| 12:46.07 | eXcAliBuR | :/ |
| 12:46.13 | eXcAliBuR | thanks |
| 12:46.23 | MrTelephone | check your tftp logs to make sure it doesn't say access denied when the phone tries to save the file |
| 12:48.54 | aberrios | bangs head on desk. "So, you've plugged your ethernet cable into the RJ11 phone line? And now your broadband doesn't work?" |
| 12:49.11 | MrTelephone | for real? |
| 12:49.16 | aberrios | uh-huh |
| 12:49.24 | MrTelephone | talk about over doing the voltage by about 5000% |
| 12:49.36 | Chainsaw | I can only see that working with ISDN on RJ45. |
| 12:49.40 | aberrios | get the reply tho |
| 12:49.53 | aberrios | "Well the cable was just dangling around so I thought I'd best plug it in somewhere" |
| 12:49.56 | Chainsaw | But yes, the NIC wouldn't like the 48V that results. |
| 12:50.14 | MrTelephone | plug your phone line into your NIC and wait for a call |
| 12:50.26 | jacc0 | lol |
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| 12:53.47 | MrTelephone | anyone here with an atlas 550? |
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| 13:11.56 | makmak78 | hello! anybody with some knowledge about calls getting dropped in mid conversation |
| 13:12.19 | makmak78 | and possible solution |
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| 13:18.06 | x1user | Hi, can anybody explain me what is group in mobile.conf? |
| 13:20.11 | kaldemar | x1user: you mean chan_mobile.conf? |
| 13:20.43 | x1user | Yes, mine is called mobile.conf. |
| 13:21.50 | kaldemar | you can dial by group, using multiple devices. Dial(Mobile/g1/12345) would dial using the first available device that belongs to group 1. |
| 13:27.24 | bougyman | vicidial question: can it run on newer asterisk than the 1.4.21.2 they want it on? |
| 13:27.31 | bougyman | i can't get zaptel to compile on our kernel. |
| 13:27.44 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 13:28.01 | jacc0 | @bougyman: use dahdi |
| 13:28.10 | bougyman | jacc0: for vicidial? |
| 13:28.22 | bougyman | i'm looking for a known good version |
| 13:28.33 | bougyman | i've had hell getting vici to work with versions that it's not built for. |
| 13:29.27 | jacc0 | okay, i'm sure this is the right channel for vicidail questions |
| 13:29.54 | jacc0 | try #vicidial |
| 13:30.35 | bougyman | k |
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| 13:30.46 | *** join/#asterisk bougyman (bougyman@pdpc/supporter/gold/bougyman) |
| 13:30.56 | bougyman | hah, one bot and chanserv in #vicidial |
| 13:31.09 | bougyman | mind if I lurk waiting for a user? |
| 13:31.26 | jacc0 | i don't |
| 13:31.38 | aberrios | bougyman, try the linkedin vicidial groups |
| 13:31.45 | aberrios | i think they're active |
| 13:32.11 | bougyman | aberrios: i'm hoping the vici author shows up to cluecon |
| 13:32.14 | bougyman | he was here last year. |
| 13:32.25 | bougyman | then I can get the info from the horses mouth. |
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| 14:17.58 | raden | Katty, MORNING :D |
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| 14:43.57 | *** join/#asterisk tamiel (~tamiel@65.204.112.138) |
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| 14:53.40 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 14:56.38 | hobodave | hey guys, I have a couple of SIP for dummies questions that I was hoping you guys could help me with. We have IP phones through our ISP, which hosts our PBX. We don't have any special hardware in our office that I'm aware of besides the phones (Polycom SoundPoint 501) a router with QoS and a switch. I am very comfortable with networking protocols and engineering outside of SIP, but I'm curious how |
| 14:56.38 | hobodave | <PROTECTED> |
| 14:56.40 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:57.05 | coppice | all I ever got from a horse's mouth was bad breath |
| 15:00.54 | p3nguin | hobodave: VoIP uses UDP. |
| 15:00.56 | tully` | is there a way to make asterisk start an EAGI() function but not wait for completion of the script? It's calling a php script that's piping the data over to another box. The problem I'm having is that asterisk isn't detecting a hangup when this happens properly. |
| 15:01.20 | hobodave | p3nguin ah, thanks I didn't know that. That makes sense now obviously, duh. |
| 15:01.52 | hobodave | p3nguin that still is just routed over the webs as a destination IP and port, how does it get to the right phone once it hits my network? |
| 15:02.01 | p3nguin | SIP traditionally uses UDP, but there are some UAs that are using SIP over TCP. RTP (the media stream) is UDP, though. |
| 15:02.21 | Qwell | coppice: strangely, I just learned something, because of that comment. |
| 15:02.25 | p3nguin | I'd guess that your NAT works well. |
| 15:02.42 | Qwell | coppice: The reason you don't look a gift(ed) horse in the mouth, is because you can tell a horses age from its teeth, and doing so would be considered rude. |
| 15:03.10 | hobodave | p3nguin that doesn't make sense though, all the packets would be coming in with a dest port of 5060, how would it know which phone to route that to? |
| 15:03.16 | coppice | Qwell: this much I know, but how did you get that from my comment? |
| 15:03.21 | hobodave | does it do some Layer 7 inspection? |
| 15:03.41 | Qwell | coppice: wikipedia. I was going to attempt to be clever, and find something else you could obtain from a horses mouth. |
| 15:04.12 | serafie | wonders how a horse's mouth affects coppice's breath. |
| 15:04.25 | n3hxs | I always thought that you shouldn't look as you would find out what kind of "gift" you really got. |
| 15:04.30 | Qwell | serafie: +1, semantics |
| 15:05.47 | p3nguin | hobodave: I don't really know how to explain the internal workings of it, but your phones send packets saying how to be reached. When there's a call for it, asterisk sends the call to the location. It is interesting to think about now that you have brought it up. |
| 15:06.01 | serafie | n3hxs: I have always interpretted it Qwell's way, as in: You've gotten a gift, don't be ungrateful if it's an old weak one. |
| 15:06.48 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:06.48 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:06.50 | Qwell | Worst case scenario: STEW! |
| 15:06.58 | serafie | and GLUE! |
| 15:07.01 | Qwell | touche |
| 15:07.07 | Qwell | also, those rhyme. |
| 15:07.14 | serafie | TRUE. |
| 15:07.14 | aberrios | hobodave, p3nguin, iirc some NATing rewrites the destination in the SIP packet headers and only in SIP packet headers. |
| 15:07.21 | coppice | I think the idea is silly. some gifts come with major liabilities |
| 15:07.59 | n3hxs | right serafie, and this day I have learned another meaning. |
| 15:09.06 | hobodave | ahh I think I found something p3nguin and aberrios, it looks like there is something called the SIP Application-level Gateway (ALG) which "most routers implement". So yes, it is a Layer 7 protocol |
| 15:09.37 | p3nguin | If your router is using ALG, it'll break Asterisk's ability to reach the NATed phones. |
| 15:09.57 | p3nguin | Rule number one: always disable any SIP fix-up on your NAT device. |
| 15:10.02 | hobodave | p3nguin well then I don't know how our phones are reached :) |
| 15:10.13 | aberrios | magic.. SHAZAM! |
| 15:10.20 | hobodave | they're definitely NAT'd though, and I'm assuming our provider is using Asterisk on their side |
| 15:11.34 | p3nguin | The phones have a private address that asterisk still sees, even though the public address is where the calls have to be sent. |
| 15:13.44 | *** join/#asterisk sassyn (~sassyn@bzq-82-80-242-217.cablep.bezeqint.net) |
| 15:13.48 | sassyn | hi all |
| 15:13.56 | sassyn | some quick question |
| 15:14.25 | hobodave | p3nguin ok so it must be layer 7, but the ALG isn't the thing doing it? |
| 15:14.37 | sassyn | in 1.4 source tree I used spandsp to have the fax_app.so application |
| 15:14.50 | sassyn | sorry app_fax... to support NVFAX |
| 15:14.51 | hobodave | there has to be some logic in the router that knows how to handle the SIP protocol |
| 15:14.54 | p3nguin | It's probably more like a combination of layer 4, 6, and 7. |
| 15:15.02 | sassyn | what is the status of it in version 1.6 |
| 15:15.25 | Qwell | sassyn: Use 1.8, and switch to res_fax_spandsp |
| 15:15.54 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 15:16.11 | sassyn | Qwell, is it stable 1.8? I used only 1.4 for almost 5 years now |
| 15:16.28 | Qwell | We wouldn't have released it if we didn't think it was usable. |
| 15:16.39 | sassyn | also Not sure FreePBX support 1.8 |
| 15:16.46 | Qwell | FreePBX supports 1.8 |
| 15:16.51 | sassyn | OK thanks |
| 15:17.47 | sassyn | Qwell, can u just tell me what a little bit more history about the fax_app |
| 15:18.07 | Qwell | No, I know nothing about it. |
| 15:18.29 | sassyn | what is now the status, cause I see asterisk in version 1.6 has it's won fax_app implemnation |
| 15:18.36 | p3nguin | its |
| 15:18.38 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:18.40 | p3nguin | not it's |
| 15:18.49 | sassyn | which wasn't in version 1.4 only after patching |
| 15:19.18 | sassyn | so my question is if in version 1.6 the fax is like NVFAX? |
| 15:19.48 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
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| 15:35.59 | cerienjean | hobodave.... |
| 15:36.09 | hobodave | cerienjean yes? |
| 15:36.10 | cerienjean | I've just browsed through that and it is more simple |
| 15:36.40 | cerienjean | the phone on power up sends a registration FROM a given port (random usually) to your hosted pbx, dest port 5060 |
| 15:37.14 | cerienjean | so you have a dialog between port <random> and port 5060 |
| 15:37.50 | cerienjean | so when the pbx wants to send sip to your phone, it sends from 5060 to the port your phone used, and your nat router just passes it on |
| 15:38.01 | hobodave | cerienjean yea I was reading something similar, but it apparently is more complex than that based on the type of NAT implemented |
| 15:38.20 | cerienjean | I think the nat is just about the ip@, not the port |
| 15:38.27 | hobodave | full cone, restricted cone, port restricted cone, and symmetric NAT |
| 15:38.32 | cerienjean | yes |
| 15:38.42 | p3nguin | According to the RFC, traffic should be sent to port 5060, not the source port that the other traffic came from. |
| 15:38.45 | cerienjean | but mainly, it is about finding the right ip |
| 15:39.04 | hobodave | there is still some Layer 7 inspection done though |
| 15:39.07 | cerienjean | I dont know the RFC well enough |
| 15:39.11 | *** join/#asterisk jpcansa (~JP@201.196.184.142) |
| 15:39.21 | cerienjean | but I tell you, real life is what happens ! |
| 15:39.43 | p3nguin | When NATed, the private address is still contained in the packets that Asterisk receives. |
| 15:40.01 | p3nguin | I think that will be used when sending traffic back. |
| 15:40.17 | cerienjean | that is when the nat parameters of asterisk come into play |
| 15:40.25 | p3nguin | exactly |
| 15:40.46 | cerienjean | when it will replace the ip @ with the one it actually received the packet from |
| 15:41.00 | cerienjean | but still uses the port it received the packet from |
| 15:41.25 | cerienjean | if you put a wireshark on a replicated port of your switch, you'll see that |
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| 15:46.13 | lirakis | hey anyone ever used a yealink t38 phone? We are testing some here and the rfc2833 dtmf doesnt seem to be recognized by asterisk, but it is by freeswitch |
| 15:46.22 | lirakis | i was curious if any one had any experience |
| 15:47.16 | p3nguin | Sometimes inband dtmf works in situations where normal (rfc2833) doesn't. |
| 15:48.14 | lirakis | yeah .. im not trying to use the phone .. .just testing it out b/c the mfg sent to us |
| 15:48.39 | lirakis | i see that it sends the first rtpevent packet with marker=true and event duration 160 |
| 15:48.51 | lirakis | then the second one it sends has event duration of 80 |
| 15:48.57 | lirakis | then it goes up consistently |
| 15:49.01 | lirakis | which .. is odd |
| 15:49.21 | lirakis | i assume the event duration should continue to increment up to a higher value with each packet |
| 15:49.43 | lirakis | i tested a polycom and it was continuously incrementing from first marker to the end |
| 15:50.43 | lirakis | i wasnt sure if any one knew if having a .. sort of "out of order" event duration for 2833 dtmf would cause it not to be recognized |
| 15:51.00 | lirakis | the only other difference is the yealink sends volume 13 and the polycom sends volume 15 |
| 15:51.29 | lirakis | (that i am able to tell) |
| 15:52.03 | *** join/#asterisk Docfxit (~Console1@netblock-75-79-6-10.dslextreme.com) |
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| 15:53.02 | Victor_Yure | does anybody knows a way to use "includes" with asterisk realtime (dynamic)? |
| 15:54.00 | Docfxit | A long time ago I found a file that stored the voicemail passwords in Asterisk. I don't remember what file that is. Can someone please let me know the name and path of the file? |
| 15:54.29 | p3nguin | docfxit: voicemail.conf |
| 15:56.25 | pabelanger | Victor_Yure: can't |
| 15:56.29 | pabelanger | not supported |
| 16:01.17 | dexteruk | How can i error handle a non-existent destination for gosub rather than it just stopping |
| 16:05.47 | dexteruk | I dont want the dialplan to stop when i cannot find destination |
| 16:06.51 | atheos | dexterous use the invalid extension |
| 16:08.00 | p3nguin | Gosub() is trying to go to an extension that doesn't exist? |
| 16:08.01 | *** join/#asterisk eduardonunesp (~eduardo@200.175.61.226) |
| 16:09.06 | eduardonunesp | there is a way to put \n (line feed) in extensios.conf ? |
| 16:09.16 | p3nguin | Why would you need to? |
| 16:11.52 | eduardonunesp | to send a string by SMS |
| 16:12.18 | eduardonunesp | the asterisk put a another slash when pass by her functions |
| 16:12.52 | eduardonunesp | then the double slash (\\) are inserted in the string |
| 16:17.26 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 16:21.53 | Docfxit | p3nguin: I don't see any passwords in voicemail.conf |
| 16:22.01 | p3nguin | Scroll down. |
| 16:22.48 | Docfxit | p3nguin: I've looked at all the records in the entire file. |
| 16:23.54 | p3nguin | Do you see the voice mail contexts there? |
| 16:25.02 | *** join/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com) |
| 16:28.08 | Docfxit | p3nguin: No. I did a search for contexts and I didn't find voice mail contexts |
| 16:28.33 | p3nguin | The contexts are not named "context," so it's no wonder your search didn't find any. |
| 16:28.51 | p3nguin | Look for [default] at least. That's the default voicemail context. |
| 16:30.25 | Docfxit | p3nguin: I have a default section. There is only one line that isn't commented out. It's an example. |
| 16:30.41 | p3nguin | Do you have any OTHER contexts? |
| 16:31.31 | Docfxit | p3nguin: Yes. It also has only one line that isn't commented out. It's an example also. |
| 16:32.17 | p3nguin | So what you're telling me is that you don't have any voice mail stuff configured. |
| 16:32.26 | p3nguin | Either that, or you're not using pure asterisk. |
| 16:32.41 | p3nguin | I'm leaning toward the latter, given how familiar you are with all of it. |
| 16:33.33 | Docfxit | p3nguin: There are lots of voicemail boxes configured. I just don't see them in this file. |
| 16:35.19 | p3nguin | So... are you using FreePBX or Asterisk GUI? |
| 16:38.32 | Docfxit | p3nguin: I am not using FreePBX, Right now I'm looking at the files in gnome |
| 16:39.35 | p3nguin | Any sane asterisk admin would have configured voice mail boxes in voicemail.conf. If you don't have yours there, that indicates to me that something like FreePBX has configured the "users" in another file. |
| 16:40.00 | p3nguin | I'll give you a hint... |
| 16:40.02 | p3nguin | ~users.conf |
| 16:40.02 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 16:42.39 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 16:43.30 | *** part/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 16:44.41 | Jcook_5xData | Can someone take at look at this http://pastebin.com/hzck9bxh and see what I did wrong. The calls are not rolling to the next step on timeout |
| 16:45.03 | eduardonunesp | ok there is no way to put \n in dialplan |
| 16:46.07 | p3nguin | jcook_5xdata: Are you using Asterisk 1.2? |
| 16:46.17 | Jcook_5xData | no 1.6 |
| 16:46.20 | p3nguin | jcook_5xdata: Convert your pipes to commas. |
| 16:46.23 | atheos | eduardonunesp - test from the manager interface. if it works, use an AGI script thank can forward the request to the manager. |
| 16:46.47 | *** join/#asterisk mhaddog (~mhaddog@z65-50-116-231.ips.direcpath.com) |
| 16:46.48 | Jcook_5xData | simple enuff |
| 16:46.59 | Jcook_5xData | thanks |
| 16:47.03 | p3nguin | jcook_5xdata: And I'm also curious if you intend for callERs to be able to do DTMF transfers or not. |
| 16:47.15 | eduardonunesp | atheos, there is a great problem, i want simple dialplan but ok |
| 16:47.22 | Docfxit | p3nguin: I didn't originally configure this box. I am going to build a new box and I'll be looking for someone that can convert everything from this box to FreePBX. Hopfully I'll find someone that knows what they are doing. |
| 16:47.45 | p3nguin | Someone who knows what he is doing probably wouldn't use FreePBX. |
| 16:47.56 | billmania | When I use the feature codes to BLIND or ATTENDED transfer a call which came in through a queue on asterisk 1.8.4.2, from the answering queue member to another queue member, the call is transferred but the queue still lists the original member as "In use". Also, No TRANSFER entry is written to the queue_log. How do I correct this? |
| 16:47.59 | p3nguin | Just sayin'. |
| 16:47.59 | atheos | eduardonunesp I can understand that desire |
| 16:48.23 | billmania | My queue members are Local/5101@from-internal/nb. |
| 16:48.59 | eduardonunesp | atheos, exten => 9991, 1, System(prog "blablablaba \n blabalba") |
| 16:49.18 | billmania | I've noticed this after the transfer has happened: |
| 16:49.18 | billmania | Local/5101@from-inte from-internal 800 1 Ringing AppQueue (Outgoing Line) 5102 |
| 16:49.44 | p3nguin | jcook_5xdata: I doubt that's what you really intend, so you shouldn't have T as a Queue() option. You really shouldn't even use t if your agents have SIP phones (SIP phones have transfer buttons). |
| 16:50.06 | Docfxit | p3nguin: I heard that was the most poplular one. What should I use in the new box? |
| 16:50.11 | atheos | eduardonunesp - that, or using AGI() - I just send everything to Perl Asterisk::AGI |
| 16:50.16 | p3nguin | docfxit: Asterisk. |
| 16:50.58 | eduardonunesp | atheos, thanks |
| 16:51.15 | Docfxit | p3nguin: Great. |
| 16:53.09 | Docfxit | p3nguin: I just made a change in users.conf. I commented out one line vmsecret for one user. How can I reload the users.conf file? |
| 16:53.23 | p3nguin | Restart asterisk. |
| 16:53.37 | p3nguin | See infobot's comment above. |
| 16:55.13 | Jcook_5xData | p3nguin, your right I can not remember why i have that in there, I think i was old code for 1.1 or 1.2 system i reused |
| 16:57.24 | billmania | If I use the Polycom phone's soft transfer button, a TRANSFER record is written to queue_log but then neither member is listed as "In use", both AppQueue and Queue Applications disappear from "core show channels verbose" and the COMPLETE record is never written to queue_log. |
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| 17:06.22 | catphish | is there a known problem with not being able to load new musiconhold classes into a running asterisk? |
| 17:33.12 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 17:33.12 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 17:33.48 | catphish | Docfxit: asterisk -rx "core restart now" |
| 17:33.58 | catphish | that will connect to the local console and initiate an immediate restart |
| 17:34.14 | catphish | you can initiate any command in that way |
| 17:34.20 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:34.30 | catphish | alternatively, just use an init script |
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| 17:44.04 | REdOG | anyone familliar with switchvox? |
| 17:44.53 | anonymouz666 | REdOG: No sir. Sorry. Maybe there's a channel for switchvox? |
| 17:45.06 | REdOG | yea, its quiet though |
| 17:45.30 | REdOG | it's basically an asterisk box but I can't even ssh in |
| 17:45.46 | anonymouz666 | switchvox allows that? |
| 17:46.51 | REdOG | no |
| 17:48.11 | anonymouz666 | they just put disallow=REdOG then ;) |
| 17:49.36 | _Corey_ | Uh no, you can't access the shell in Switchvox :) |
| 17:49.38 | REdOG | all my transfers have to be blind transfers or they don't transfer |
| 17:49.58 | anonymouz666 | setting up heartbeat+asterisk |
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| 18:30.12 | sassyn | Hi Again |
| 18:30.54 | sassyn | Again: One question to make some issue more clear |
| 18:31.06 | sassyn | version 1.4 didn't include any fax support |
| 18:31.18 | sassyn | only if u applay the spandsp pacthes |
| 18:31.26 | anonymouz666 | I think you are correct. |
| 18:31.37 | anonymouz666 | just T.38 pass-through |
| 18:31.44 | sassyn | which is now agx-ast-addons. |
| 18:31.49 | anonymouz666 | 1.4 for FAX really sux. |
| 18:32.05 | sassyn | 1.6 did applay some FAX support |
| 18:32.10 | anonymouz666 | use 1.8 for that |
| 18:32.10 | sassyn | t.38 |
| 18:32.22 | anonymouz666 | there's a patch for t.38 gw in 1.8 |
| 18:32.30 | anonymouz666 | fax buffer for dahdi, etc. |
| 18:32.32 | *** join/#asterisk darkskiez (~darkskiez@62-50-207-4.client.stsn.net) |
| 18:32.37 | anonymouz666 | support spandsp... |
| 18:32.51 | sassyn | so version 1.8 as two modes |
| 18:32.57 | sassyn | one spandsp |
| 18:33.03 | sassyn | and second digum support |
| 18:33.13 | sassyn | both are them for t.38 ? |
| 18:33.16 | sassyn | right? |
| 18:34.41 | p3nguin | I use 1.4 for fax. fax for asterisk seems to do okay most of the time. |
| 18:35.31 | sassyn | p3nguin, so u used the spandsp pacthes right? |
| 18:35.35 | sassyn | AKA NVFAX? |
| 18:35.45 | p3nguin | I didn't use any patches. |
| 18:36.01 | sassyn | the vanilla version of 1.4.X? |
| 18:36.28 | sassyn | or u used some rpm or deb pkg? |
| 18:37.40 | singler | for regular faxing (fax as regular device) vanilla asterisk works fine, but if you want T38 (non-pass through), you need patches |
| 18:38.33 | *** join/#asterisk neurosys (~neurosys@65.204.112.138) |
| 18:38.36 | anonymouz666 | p3nguin: you are lucky then. |
| 18:38.55 | anonymouz666 | specially multi-pages faxing doesn't work pretty, exaclty as stated in the book (3rd edition). |
| 18:38.56 | p3nguin | I don't have any t.38 stuff. I'm just doing fax over voice over ip. |
| 18:39.41 | anonymouz666 | p3nguin: I am talking about the same setup as you |
| 18:39.54 | anonymouz666 | SIP -> LAN -> ASTERISK -> PRI |
| 18:40.18 | p3nguin | It used to work correctly on my last system, but then I moved my asterisk to another computer and just reinstalled ffa the other day. I sent some faxes to it and it kept getting a timeout. I don't know if it was because of where I was sending from or if it was a problem with my configuration. |
| 18:40.44 | sassyn | singler, so 1.4 support fax? |
| 18:40.46 | p3nguin | I don't get enough faxes to get a good sample of senders. |
| 18:41.16 | sassyn | singler, what can be done. Sorry about my stupid question... but I don't get what is supported in version 1.4 without pacthes |
| 18:41.32 | singler | sassyn: what exactly do you want to do? |
| 18:41.47 | singler | if fax will be as regular phone then it should work |
| 18:41.48 | p3nguin | fax over sip using "fax for asterisk" works without any patches |
| 18:42.02 | p3nguin | Beyond that, I have no idea. |
| 18:42.40 | sassyn | so this means if I have a SIP fax machine then I can sent faxes as regular |
| 18:42.51 | sassyn | in version 1.4 without pacthes... |
| 18:42.55 | singler | yes |
| 18:43.05 | singler | you also can have analog fax via converter |
| 18:43.07 | sassyn | singler, I'm just trying to understand some general info |
| 18:43.12 | p3nguin | You can get one single fax for asterisk channel for free from digium. |
| 18:43.25 | p3nguin | Just look for "free fax for asterisk." |
| 18:43.28 | singler | general info - faxing is a regular call where machines "talk" |
| 18:43.31 | sassyn | this is called T38 pass through right? |
| 18:43.52 | singler | T38 is additional stuff, it adds reliability |
| 18:44.00 | singler | but all parties must support it |
| 18:44.17 | sassyn | ok. so what is the diff betwheen pass through and non pass through |
| 18:44.21 | singler | eg: T38 fax -> asterisk -> T38 sip provider |
| 18:45.14 | sassyn | singler, ok thanks |
| 18:45.15 | singler | in pass-through asterisk do not manage t38 (nor it creates - it is not t38 gateway), asterisk only passes t38 through it |
| 18:45.38 | sassyn | ok now I think I got it |
| 18:45.55 | singler | if for example you have PRI line, you need T38 gatewaying, so asterisk could create T38 "connection" with your fax |
| 18:46.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:46.39 | sassyn | so version 1.4 without patches can work as a t38 pass-through like regular phone where machine "talk" |
| 18:46.41 | singler | but on small lan I doubt t38 very helps, basically it can retransmit lost packets |
| 18:47.14 | singler | yes, if machines support t38 |
| 18:47.21 | sassyn | ok |
| 18:47.27 | singler | in other case it just as regular call |
| 18:47.36 | singler | think of t38 as reliability addon |
| 18:47.42 | sassyn | OK |
| 18:48.14 | sassyn | 1.4 asterisk can't get faxes and sent them to user without pacthes |
| 18:48.30 | sassyn | only if u applied spandsp - u get what it is could NVFAX |
| 18:48.56 | sassyn | if this is true I think i got it this time :-) |
| 18:49.17 | singler | I do not know about NVFAX, but you could use hylax+iaxmodem setup to receive/send faxes on server |
| 18:49.50 | sassyn | well NVFAX is the app_fax.so addon from spandsp |
| 18:50.23 | sassyn | so inoder to use iaxmodem u will habe to applay the pacthe right? |
| 18:50.32 | sassyn | have* |
| 18:50.55 | singler | no |
| 18:51.18 | sassyn | so what are the spandsp patches for? |
| 18:51.33 | singler | for native fax and t38 support |
| 18:51.47 | singler | in my case other applications handle fax |
| 18:52.08 | sassyn | but asterisk 1.4 allready have this allready - as u point out? |
| 18:52.16 | singler | iaxmodem communicates with asterisk via IAX, hylafax sees regular modem in /dev/ |
| 18:54.18 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 18:54.18 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 18:54.35 | singler | isn't it already EOL? |
| 18:54.40 | p3nguin | 1.8 branch hasn't been out THAT long yet; not everyone is interested in a migration to the new branch. |
| 18:54.45 | sassyn | singler, OK but u still can in 1.4 to do T38 pass through |
| 18:54.46 | p3nguin | It's in security only mode. |
| 18:55.26 | sassyn | but if u want asterisk to send and resive fax u need to pacth it using spandsp right? |
| 18:55.38 | singler | so regular bugs will not be fixed, if they appear after security fix? |
| 18:55.53 | sassyn | hylax+iaxmodem is just anther method to do so |
| 18:55.59 | p3nguin | NO. WTF is your obsession with patching asterisk for faxing? |
| 18:56.18 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 18:56.23 | p3nguin | There is no more development on the 1.4 branch. |
| 18:56.32 | p3nguin | Only security problems will be addressed. |
| 18:56.39 | sassyn | p3nguin, it is just I have 1000 ext asterisk 1.4.29 install on debain |
| 18:56.43 | sassyn | with FreePBX 2.9 |
| 18:56.53 | sassyn | and the NVFAX option is gray out |
| 18:57.08 | sassyn | and app_fax.so doesn't exitst |
| 18:57.34 | sassyn | so I trying to understand the issue - first why? and second how can I do that. |
| 18:57.52 | *** join/#asterisk darkskiez (~darkskiez@2001:470:9278:5:2e0:4cff:fe68:1e29) |
| 18:58.02 | sassyn | I need version 1.4.29 since lots of application we wrote support only this version at the moment |
| 18:58.44 | sassyn | and I'm kind of lost with too much documnations saying this and this - so I'm trying to get the picture |
| 18:59.12 | sassyn | sorry if I ask stupid question |
| 19:00.29 | *** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net) |
| 19:00.31 | d_preston215 | Has anybody had any issues using the leastrecent ring strategy with queues? |
| 19:00.54 | billmania | d_preston215: Yes, with asterisk 1.6. |
| 19:01.06 | sassyn | singler, |
| 19:01.10 | sassyn | still here? |
| 19:01.39 | singler | here, but I cannot help you with freepbx |
| 19:02.00 | *** join/#asterisk Tim_Toady (~moi@178.128.18.161) |
| 19:02.02 | billmania | d_preston215: That queue strategy appears to work correctly in asterisk 1.8.4.2. |
| 19:02.41 | sassyn | singler, np |
| 19:03.00 | sassyn | maybe I will try to install 1.8 |
| 19:03.08 | sassyn | but does it include fax support? |
| 19:03.12 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 19:03.25 | sassyn | where user can get fax to email or send fax to asterisk? |
| 19:04.13 | singler | sassyn: I think if you have 1.4 setup you should think if it really worth upgrade |
| 19:04.22 | singler | especially with freepbx |
| 19:04.32 | p3nguin | I have no problem with fax to email in 1.4. |
| 19:04.49 | p3nguin | I use fax for asterisk, tiff2pdf, and mutt. |
| 19:05.10 | sassyn | p3nguin, do u have the app_fax.so module? |
| 19:05.19 | p3nguin | Fax comes in, and is saved as a tif. tiff2pdf converts it to pdf. mutt mails the pdf to me. |
| 19:05.31 | p3nguin | No, I do not have that. I use fax for asterisk. |
| 19:05.42 | sassyn | what is fax for asterisk? |
| 19:06.06 | p3nguin | infobot: ffa |
| 19:06.06 | infobot | Free For All |
| 19:06.09 | p3nguin | heh |
| 19:06.16 | p3nguin | Fax For Asterisk. Look it up. |
| 19:06.19 | p3nguin | digium.com |
| 19:06.49 | p3nguin | infobot: ffa is also Fax For Asterisk |
| 19:06.49 | infobot | p3nguin: okay |
| 19:06.54 | p3nguin | ~ffa |
| 19:06.54 | infobot | Free For All. Fax For Asterisk |
| 19:07.01 | d_preston215 | So basically leastrecent is a crapshoot in asterisk 1.6? |
| 19:10.17 | *** join/#asterisk darkskiez (~darkskiez@2001:470:9278:5:2e0:4cff:fe68:1e29) |
| 19:12.25 | billmania | d_preston215: I'll state that it actually doesn't work correctly in 1.6. |
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| 19:17.25 | billmania | I'm having trouble transferred a queue call in 1.8.4.2. |
| 19:18.00 | billmania | After the call is transferred, the AppQueue application thinks the the transfer is still in progress. |
| 19:18.21 | *** join/#asterisk zyphlar (~AndChat@206.29.182.215) |
| 19:18.27 | billmania | I think because of that it thinks the original queue member is still in use and the recipient of the transfer is not in use. |
| 19:18.51 | billmania | My queue members are Local/5102@from-internal/nb. What am I missing? |
| 19:19.37 | zyphlar | Anyone know of a good cdr for reporting dropped and voicemail calls? |
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| 19:49.42 | eduardonunesp | so i just try many ways to line break at asterisk within extensions.conf |
| 19:49.47 | eduardonunesp | so found the way |
| 19:50.15 | eduardonunesp | Set(LF=${SHELL(echo)}) |
| 19:50.48 | eduardonunesp | System(Text text text ${LF}) |
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| 19:51.14 | Qwell | eduardonunesp: oh my |
| 19:51.38 | Qwell | leifmadsen: ^^^ |
| 19:51.45 | jeffspeff | how can i get the parkinglot to announce the parked exten on the phones display? |
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| 19:59.09 | paulc | jeffspeff: what version of Asterisk are you using? |
| 19:59.23 | jeffspeff | paulc, 1.8.5 |
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| 19:59.40 | jeffspeff | paulc, also, i'm using cisco spa504g phones, all is sip |
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| 20:00.30 | paulc | jeffspeff: I'm not at my regular desk right now but I'm sure I read something about that in one of the sample/example config files.. the phone has to support connected line updating though.. I don't know if the SPA's do (but know that Polycom do) |
| 20:02.30 | leifmadsen | Qwell: well isn't that an interesting way to do it :) |
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| 20:54.06 | Jcook_5xData | I need to test some sip phone what a good beside Polycom? |
| 20:54.28 | Jcook_5xData | linksys, snom? |
| 20:55.34 | paulc | Polycom are great.. Linksys are "alright".. The Aastra's aren't bad.. I haven't played with the Yealink but I've heard mixed reports. It's on my list of things to do. |
| 20:55.39 | beek | You mean there's a SIP phone other than Polycom? |
| 20:55.49 | Jcook_5xData | lol |
| 20:56.14 | Jcook_5xData | I bought 20 Ploycom 650 but we are getting a bad echo in the handset |
| 20:56.29 | beek | Jcook_5xData: when calling what? |
| 20:56.55 | *** join/#asterisk darkskiez (~darkskiez@2001:470:9278:5:2e0:4cff:fe68:1e29) |
| 20:56.59 | *** join/#asterisk thehar (thehar@diddlebox.thehar.com) |
| 20:57.26 | Jcook_5xData | most outside call but it real bad on long distant |
| 20:58.05 | Jcook_5xData | What make me think it the phone most of us use Plantronics headset we get no echo it just when you use handset |
| 20:58.47 | Jcook_5xData | And polycom no help unless you have some code |
| 20:59.35 | p3nguin | The most common reason for a SIP phone to echo is having the handset volume too high. |
| 20:59.46 | p3nguin | (or when calling to an analog technology that has echo) |
| 21:00.46 | Jcook_5xData | what funny is I switch out the handset for old Nortel 1980's handset and no echo |
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| 21:02.39 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:06.10 | Jcook_5xData | p3nguin, I agree with you but the fact that we have no echo with headset or on the old handset. |
| 21:07.13 | Jcook_5xData | oohh well, today is done. I live to fight it tomorrow. thank for the help :) |
| 21:11.17 | wonderworld | hi, i used /etc/asterisk/asterisk.conf to set astagidir to a custom dir. i restarted asterisk but it still epxpects agi-scripts to reside in /usr/share/asterisk/agi-bin/ . what might be wrong? tnx |
| 21:13.58 | chazzam | wonderworld: did you leave the "(!)" after the "[directories]" definition? |
| 21:14.05 | wonderworld | never mind, i should have read the first line of the file |
| 21:14.06 | chazzam | if so, then it won't get read by Asterisk |
| 21:14.11 | chazzam | =p |
| 21:14.14 | wonderworld | yeah, just saw it |
| 21:14.15 | wonderworld | thanks |
| 21:14.17 | wonderworld | ;) |
| 21:14.27 | chazzam | got burned by that one too |
| 21:14.35 | *** part/#asterisk kraptv (ryan@mail.skylab.org) |
| 21:15.18 | wonderworld | <PROTECTED> |
| 21:39.20 | jeffspeff | I have multiple numbers coming from the same SIP provider... how can I control which context those numbers go to when the call comes into asterisk? |
| 21:39.50 | chazzam | do you have multiple account with the sip provider? |
| 21:39.59 | jeffspeff | no, same account |
| 21:40.17 | chazzam | then... all you could do would be match the incoming did, all in the same context |
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| 21:41.22 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 22:00.26 | Rufus | go...went...gone... |
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| 22:04.57 | Micc | Anyone got any ideas why I can't get the gmt offset to work on a polycom 501? Setting it in the cfg file and locally on the phone seems to have no affect. I'm leaning toward bad firmware at this point. |
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| 22:19.00 | timeshell | Greetings. Since switching from 1.6.0.26 to 1.8.x voicemails have not been getting emailed to their users. Any ideas? |
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| 22:22.02 | Maliuta | timeshell: I had a similar set of issues in my 1.6->1.8 upgrade. I ended up doing a complete re-install and rebuilding my configs one at a time |
| 22:22.18 | timeshell | I, |
| 22:22.20 | timeshell | Um |
| 22:22.25 | timeshell | That's not really a solution |
| 22:25.21 | Maliuta | well it was the only thing that fixed my issue with my extensions not working properly (things like time based goto's just plain not working, problems with voicemail and pass through between SIP and DAHDI channels) |
| 22:26.00 | Maliuta | most of my old confs slipped in to the new install with no real change, some needed a mild tweek |
| 22:27.12 | Maliuta | timeshell: you _should_ be able to do it in a test environment and then push the working configs to the production environment and simply restart/reload the confs with minimal interruption |
| 22:37.00 | p3nguin | timeshell: Check for changes in voicemail.conf in the new version. Check the sample file that came with it. |
| 22:49.52 | timeshell | p3nguin There's no real significant difference that I can discern |
| 22:52.57 | p3nguin | I'd probably sdiff it. Maybe there's a very subtle change that causes your emails to no longer go out. |
| 22:53.07 | p3nguin | Let me see if I can figure it out. |
| 22:54.57 | p3nguin | What is your mailcmd in the conf of the current version? |
| 22:54.59 | p3nguin | the one that isn't working. |
| 22:55.14 | timeshell | Same as in the sample |
| 22:55.49 | timeshell | commented out, just like sampel |
| 22:55.51 | p3nguin | ;mailcmd ... |
| 22:55.58 | p3nguin | okay |
| 22:55.58 | timeshell | yes |
| 22:56.10 | p3nguin | Check that you have /usr/bin/sendmail on your system. |
| 22:56.35 | p3nguin | or /usr/sbin/sendmail, as it may be. |
| 22:56.50 | timeshell | confirmed. |
| 22:56.59 | timeshell | Worked in 1.6.0.26... |
| 22:57.05 | timeshell | Stopped working after upgrade. |
| 22:57.17 | p3nguin | But it doesn't work now, so we have to look at everything involved. |
| 22:57.38 | timeshell | sure |
| 22:57.39 | timeshell | ok |
| 22:58.34 | p3nguin | Did you check the bug tracker for a similar problem? |
| 22:58.44 | timeshell | Been googling. |
| 22:58.50 | timeshell | Not directly the bug tracker. |
| 22:59.01 | timeshell | Found something similar, but didn't provide resolution |
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| 23:54.16 | pdtpatrick | What is the best way to read vxml files on asterisk? Can Festival do this? we were using openvxi but its crapped out since the update to gcc |